Administering Avaya IP Office using

Manager

Release 12.0
Issue 51.1.2
June 2024
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 Contents

Part 1: Introduction........................................................................................................... 32
Chapter 1: Purpose........................................................................................................... 33
New in IP Office Release 12.0.......................................................................................... 33
Chapter 2: IP Office Manager Overview.......................................................................... 35
IP Office Manager Modes................................................................................................. 35
Security Configuration Mode............................................................................................ 36
Standard Mode Configuration Mode.................................................................................. 37
Server Edition Configuration Mode.................................................................................... 39
Shell Server Mode........................................................................................................... 40
Backward Compatibility.................................................................................................... 41
Chapter 3: Getting Started............................................................................................... 42
IP Office Manager PC requirements.................................................................................. 42
Installing the IP Office Admin Suite................................................................................... 43
Downloading Manager Admin Lite.................................................................................... 45
Starting Manager............................................................................................................. 45
Opening a Configuration.................................................................................................. 46
Login messages.............................................................................................................. 48
Changing the Manager Language..................................................................................... 50
Chapter 4: Manager User Interface................................................................................. 52
Title Bar.......................................................................................................................... 52
Toolbars......................................................................................................................... 52
The Main Toolbar...................................................................................................... 53
The Navigation Toolbar.............................................................................................. 54
The Details Toolbar.................................................................................................... 54
The Navigation Pane....................................................................................................... 54
Expanding and Collapsing the Navigation Tree............................................................ 55
The Group Pane.............................................................................................................. 55
Sorting the List.......................................................................................................... 55
Customizing the Columns Displayed........................................................................... 56
Changing the Column Widths..................................................................................... 56
Adding a New Record................................................................................................ 56
Deleting an Record.................................................................................................... 56
Validating an Record.................................................................................................. 57
Show in Groups........................................................................................................ 57
The Details Pane............................................................................................................. 57
Managing Records.................................................................................................... 58
The Error Pane............................................................................................................... 59
Altering the Automatic Validation Settings.................................................................... 59
Revalidating Configuration Settings............................................................................ 60

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 Viewing an Error or Warning....................................................................................... 60
The Status Bar................................................................................................................ 60
Configuring the Interface.................................................................................................. 61
Resizing the Manager Window................................................................................... 61
Moving the Border Between the Panes....................................................................... 62
Showing or Hiding Toolbars........................................................................................ 62
Moving Toolbars........................................................................................................ 62
Showing or Hiding Panes........................................................................................... 62
Changing the Position of the Details Pane................................................................... 63
Changing the Size of Configuration Icons.................................................................... 63
Changing Tab Display................................................................................................ 63
Chapter 5: Working with the Server Edition Manager User Interface.......................... 65
Server Edition Solution View............................................................................................ 65
Displaying the Server Edition Solution View................................................................. 66
The Solution View Network Display............................................................................. 66
Opening additional applications.................................................................................. 67
Setting all nodes to Subscription mode........................................................................ 68
Setting all nodes to Select mode................................................................................. 68
Setting the License Source in Server Edition View....................................................... 68
Adding a Server in the Server Edition Solution View..................................................... 69
System Inventories.......................................................................................................... 69
Default Settings............................................................................................................... 69
Record Consolidation...................................................................................................... 70
Telephone Features Supported Across Server Edition and SCN Networks........................... 71
Part 2: Menu Bar Commands......................................................................................... 74
Chapter 6: Menu Bar Commands.................................................................................... 75
Chapter 7: File Menu......................................................................................................... 76
File > Open Configuration................................................................................................ 76
File > Close Configuration................................................................................................ 77
File > Save Configuration................................................................................................. 77
Saving Server Edition Configuration Changes.............................................................. 77
File > Save Configuration As............................................................................................ 78
File > Change Working Directory...................................................................................... 79
File > Preferences........................................................................................................... 81
File > Offline................................................................................................................... 81
File > Advanced.............................................................................................................. 82
File > Backup/Restore..................................................................................................... 82
File > Import/Export......................................................................................................... 82
File > Exit....................................................................................................................... 82
Chapter 8: File > Preferences.......................................................................................... 84
Preferences.................................................................................................................... 84
Directories...................................................................................................................... 86

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 Discovery....................................................................................................................... 88
Visual Preferences.......................................................................................................... 89
Security.......................................................................................................................... 90
Validation........................................................................................................................ 92
Chapter 9: File > Offline................................................................................................... 94
Create New Config.......................................................................................................... 94
Open File........................................................................................................................ 94
Open File Set.................................................................................................................. 95
Send Config.................................................................................................................... 95
Receive Config................................................................................................................ 95
Chapter 10: File > Advanced............................................................................................ 96
Erase Configuration......................................................................................................... 96
Reboot........................................................................................................................... 97
System Shutdown........................................................................................................... 97
Upgrade......................................................................................................................... 98
Searching for Systems............................................................................................. 100
Changing the .bin File Directory................................................................................ 100
Change Mode............................................................................................................... 101
Audit Trail..................................................................................................................... 101
Security Settings........................................................................................................... 102
Erase Security Settings (Default).................................................................................... 102
Embedded File Management.......................................................................................... 103
Format IP Office SD Card.............................................................................................. 103
Formatting the SD card............................................................................................ 104
Recreate IP Office SD Card........................................................................................... 105
Recreating the IP Office SD Card............................................................................. 105
Memory Card Command................................................................................................ 106
Launch Voicemail Pro.................................................................................................... 107
System Status............................................................................................................... 107
LVM Greeting Utility....................................................................................................... 107
Generate WebLM ID...................................................................................................... 107
Initial Configuration........................................................................................................ 108
Add/Display VM Locals.................................................................................................. 112
Chapter 11: View Menu................................................................................................... 114
Chapter 12: Tools Menu.................................................................................................. 115
Tools > Extension Renumber......................................................................................... 115
Tools > Line Renumber................................................................................................. 115
Tools > Connect To....................................................................................................... 116
Tools > Export > User.................................................................................................... 116
Tools > SCN Service User Management......................................................................... 117
Tools > Busy on Held Validation..................................................................................... 118
Tools > MSN Configuration............................................................................................ 118
Tools > Print Button Labels............................................................................................ 118

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 Tools > Import Templates............................................................................................... 119
File > Advanced > Generate WebLM ID.......................................................................... 120
Tools > License Migration............................................................................................... 120
Chapter 13: Security Mode Menus................................................................................ 122
Chapter 14: Embedded File Management Menu.......................................................... 123
Part 3: Security Configuration.................................................................................... 126
Chapter 15: Security Administration............................................................................. 127
Service Users, Roles, and Rights Groups........................................................................ 127
Default Service Users and Rights Groups........................................................................ 129
Default Rights Groups................................................................................................... 130
Access Control.............................................................................................................. 132
Encryption.................................................................................................................... 133
Message Authentication................................................................................................. 134
Certificates................................................................................................................... 135
Implementing Security................................................................................................... 135
SRTP........................................................................................................................... 137
Chapter 16: Editing IP Office Security Settings in Manager....................................... 139
Loading Security Settings............................................................................................... 139
Saving Security Settings................................................................................................ 140
Resetting a System's Security Settings........................................................................... 140
Chapter 17: Security Mode Field Descriptions............................................................ 141
General Security Settings............................................................................................... 142
General.................................................................................................................. 142
System......................................................................................................................... 146
System Details........................................................................................................ 147
Unsecured Interfaces............................................................................................... 149
Certificates............................................................................................................. 150
Services....................................................................................................................... 158
Rights Groups............................................................................................................... 160
Group Details.......................................................................................................... 161
Configuration.......................................................................................................... 161
Security Administrator.............................................................................................. 162
System Status......................................................................................................... 163
Telephony APIs....................................................................................................... 163
HTTP..................................................................................................................... 164
Web Services.......................................................................................................... 164
External.................................................................................................................. 166
Service Users............................................................................................................... 167
Part 4: Editing Configuration Settings..................................................................... 168
Chapter 18: Editing Configuration Settings................................................................. 169
IP500 V2 Configuration Operation.................................................................................. 169
Mergeable Settings........................................................................................................ 171

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 Configuration Size......................................................................................................... 175
Setting the Discovery Addresses.................................................................................... 176
Opening a Configuration from a System.......................................................................... 177
Opening a Configuration Stored on PC........................................................................... 180
Creating an Offline Configuration.................................................................................... 180
Copying and Pasting..................................................................................................... 182
Saving a Configuration onto PC...................................................................................... 182
Sending an Individual Configuration................................................................................ 183
Sending Multiple Configurations..................................................................................... 184
Erasing the Configuration............................................................................................... 185
Default Settings............................................................................................................. 186
Chapter 19: Creating New Records............................................................................... 188
Adding a New Record Using the Details Pane................................................................. 188
Adding a New Record Using the Group Pane.................................................................. 188
Adding a New Record Using the Navigation Pane............................................................ 188
Other record creation methods....................................................................................... 189
Chapter 20: Importing and Exporting Settings............................................................ 190
Exporting Settings......................................................................................................... 191
Importing Settings......................................................................................................... 192
Chapter 21: Known System Discovery......................................................................... 193
Configuring Manager for Known System Discovery.......................................................... 193
Using Known System Discovery..................................................................................... 194
Part 5: System Configuration Settings.................................................................... 195
Chapter 22: Configuration Mode Field Descriptions................................................... 196
Configuration field display in Standard mode................................................................... 196
Configuration field display in Server Edition mode............................................................ 197
Chapter 23: BOOTP Record........................................................................................... 200
Chapter 24: Operator...................................................................................................... 202
Chapter 25: System ....................................................................................................... 203
System......................................................................................................................... 203
LAN1............................................................................................................................ 213
LAN Settings........................................................................................................... 213
VoIP....................................................................................................................... 215
Network Topology.................................................................................................... 223
DHCP Pools........................................................................................................... 228
LAN2............................................................................................................................ 230
DNS............................................................................................................................. 230
Voicemail...................................................................................................................... 232
Telephony..................................................................................................................... 239
Telephony............................................................................................................... 239
Park and Page........................................................................................................ 248
Tones and Music..................................................................................................... 249

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 Ring Tones............................................................................................................. 253
SM......................................................................................................................... 253
MS Teams.............................................................................................................. 254
Call Log.................................................................................................................. 255
TUI......................................................................................................................... 256
Directory Services......................................................................................................... 259
LDAP..................................................................................................................... 259
HTTP..................................................................................................................... 263
System Events.............................................................................................................. 265
Configuration.......................................................................................................... 265
Alarms.................................................................................................................... 266
SMTP........................................................................................................................... 271
SMDR.......................................................................................................................... 272
VCM............................................................................................................................. 274
Integrated Contact Reporter........................................................................................... 276
VoIP............................................................................................................................. 277
VoIP....................................................................................................................... 277
VoIP Security.......................................................................................................... 279
Access Control Lists................................................................................................ 282
Dialer........................................................................................................................... 282
Contact Center.............................................................................................................. 284
Remote Operations....................................................................................................... 286
Avaya Cloud Services.................................................................................................... 286
Avaya Push Notification Services.................................................................................... 289
Chapter 26: Line.............................................................................................................. 290
ACO Line...................................................................................................................... 292
ACO Line | ACO...................................................................................................... 292
ACO Line | VoIP...................................................................................................... 293
ACO Line | T38 Fax................................................................................................. 296
Analog Line................................................................................................................... 297
Line Settings........................................................................................................... 298
Analog Options....................................................................................................... 299
BRI Line....................................................................................................................... 305
BRI Line................................................................................................................. 306
Channels................................................................................................................ 310
H.323 Line.................................................................................................................... 311
VoIP Line................................................................................................................ 311
Short Codes............................................................................................................ 313
VoIP Settings.......................................................................................................... 314
IP DECT Line................................................................................................................ 317
Line | IP DECT Line................................................................................................. 317
Gateway................................................................................................................. 317
VoIP....................................................................................................................... 320

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 IP Office Line................................................................................................................ 322
Line........................................................................................................................ 322
Short Codes............................................................................................................ 327
VoIP Settings.......................................................................................................... 327
T38 Fax.................................................................................................................. 330
Legacy SIP DECT Line.................................................................................................. 331
SIP DECT Base...................................................................................................... 332
VoIP....................................................................................................................... 333
MS Teams Line............................................................................................................. 334
MS Teams.............................................................................................................. 334
VoIP....................................................................................................................... 337
Engineering............................................................................................................ 341
SIP Credentials....................................................................................................... 342
PRI Trunks.................................................................................................................... 343
E1 Line................................................................................................................... 344
E1 R2 Line.............................................................................................................. 352
T1 Line................................................................................................................... 358
T1 PRI Line............................................................................................................. 363
SIP Line....................................................................................................................... 372
SIP Line................................................................................................................. 372
Transport................................................................................................................ 376
Call Details............................................................................................................. 379
VoIP....................................................................................................................... 386
T38 Fax.................................................................................................................. 390
SIP Credentials....................................................................................................... 391
SIP Advanced......................................................................................................... 392
Engineering............................................................................................................ 398
SM Line........................................................................................................................ 400
Session Manager.................................................................................................... 400
VoIP....................................................................................................................... 403
T38 Fax.................................................................................................................. 407
S0 Line......................................................................................................................... 408
S0 Line................................................................................................................... 409
S0 Short Codes....................................................................................................... 411
Line | S0 Channels.................................................................................................. 411
Chapter 27: Control Unit................................................................................................ 413
Chapter 28: Extension.................................................................................................... 415
Extn............................................................................................................................. 416
Analog.......................................................................................................................... 419
Extension VoIP.............................................................................................................. 421
Extension H.323 VoIP.............................................................................................. 421
SIP Extension VoIP................................................................................................. 425
Extension T38 Fax........................................................................................................ 429

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 IP DECT Extension........................................................................................................ 431
SIP DECT Base Extension............................................................................................. 432
Chapter 29: User............................................................................................................. 433
User............................................................................................................................. 433
Voicemail...................................................................................................................... 442
DND............................................................................................................................. 448
Short Codes.................................................................................................................. 448
Source Numbers........................................................................................................... 449
Telephony..................................................................................................................... 450
Call Settings........................................................................................................... 450
Supervisor Settings................................................................................................. 453
Multi-line Options.................................................................................................... 457
Call Log.................................................................................................................. 459
TUI......................................................................................................................... 460
Forwarding................................................................................................................... 461
Dial In........................................................................................................................... 465
Voice Recording............................................................................................................ 465
Button Programming...................................................................................................... 466
Menu Programming....................................................................................................... 466
Huntgroup............................................................................................................... 467
4400/6400.............................................................................................................. 467
Mobility......................................................................................................................... 468
Group Memberships...................................................................................................... 472
Announcements............................................................................................................ 472
SIP............................................................................................................................... 474
Personal Directory......................................................................................................... 474
User Portal................................................................................................................... 476
Chapter 30: Group.......................................................................................................... 480
Group........................................................................................................................... 480
User List Select Members........................................................................................ 484
Queuing....................................................................................................................... 485
Overflow....................................................................................................................... 488
Fallback........................................................................................................................ 490
Voicemail...................................................................................................................... 493
Voice Recording............................................................................................................ 498
Announcements............................................................................................................ 499
SIP............................................................................................................................... 502
Chapter 31: Short Code.................................................................................................. 503
Chapter 32: Services...................................................................................................... 505
Remote Support Services.............................................................................................. 506
Service......................................................................................................................... 506
Bandwidth..................................................................................................................... 507
IP................................................................................................................................. 510

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 Autoconnect.................................................................................................................. 511
Quota........................................................................................................................... 511
PPP............................................................................................................................. 512
Fallback........................................................................................................................ 514
Dial In........................................................................................................................... 515
SSL VPN Service.......................................................................................................... 515
Service................................................................................................................... 516
Session.................................................................................................................. 516
NAPT..................................................................................................................... 517
Fallback.................................................................................................................. 518
Chapter 33: RAS.............................................................................................................. 519
PPP............................................................................................................................. 519
Chapter 34: Incoming Call Route.................................................................................. 521
Standard....................................................................................................................... 523
Voice Recording............................................................................................................ 527
Destinations.................................................................................................................. 528
Chapter 35: WAN Port..................................................................................................... 530
WAN Port..................................................................................................................... 530
Frame Relay................................................................................................................. 531
DLCIs........................................................................................................................... 532
Advanced..................................................................................................................... 533
Chapter 36: Directory Entry........................................................................................... 535
Chapter 37: Time Profile................................................................................................. 537
Chapter 38: Firewall Profile............................................................................................ 539
Firewall | Standard......................................................................................................... 539
Firewall | Custom........................................................................................................... 541
Static NAT.................................................................................................................... 543
Chapter 39: IP Route....................................................................................................... 544
IP Route | IP Route........................................................................................................ 544
RIP Dynamic Routing.................................................................................................... 545
Chapter 40: Account Code............................................................................................. 547
Account Code............................................................................................................... 548
Voice Recording............................................................................................................ 548
Chapter 41: Subscription............................................................................................... 550
Chapter 42: License........................................................................................................ 552
License......................................................................................................................... 552
Remote Server.............................................................................................................. 555
Chapter 43: Tunnel.......................................................................................................... 559
L2TP Tunnel................................................................................................................. 559
L2PT Tunnel........................................................................................................... 560
L2TP ..................................................................................................................... 561
L2TP PPP............................................................................................................... 561

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 IP Security Tunnel......................................................................................................... 562
IPSec Main............................................................................................................. 562
Tunnel | IKE Policies (IPSec).................................................................................... 563
IPSec Policies......................................................................................................... 564
Chapter 44: Auto Attendant (EVM)................................................................................ 565
Auto Attendant.............................................................................................................. 566
Actions......................................................................................................................... 568
Chapter 45: Auto Attendant (Voicemail Pro)................................................................ 571
Auto Attendant.............................................................................................................. 571
Actions......................................................................................................................... 575
Chapter 46: Authorization Codes.................................................................................. 578
Chapter 47: User Rights................................................................................................. 579
User............................................................................................................................. 579
Short Codes.................................................................................................................. 580
Button Programming...................................................................................................... 581
Telephony..................................................................................................................... 581
Call Settings........................................................................................................... 581
Supervisor Settings................................................................................................. 582
Multi-line Options.................................................................................................... 584
Call Log.................................................................................................................. 585
User Rights Membership................................................................................................ 586
Voicemail...................................................................................................................... 587
Forwarding................................................................................................................... 588
Chapter 48: Conference................................................................................................. 590
Chapter 49: ARS.............................................................................................................. 593
ARS............................................................................................................................. 593
Chapter 50: Location...................................................................................................... 598
Location....................................................................................................................... 598
Address........................................................................................................................ 601
Part 6: Configuring IP Office........................................................................................ 603
Chapter 51: Subscriptions............................................................................................. 604
Ordering Subscriptions.................................................................................................. 604
Trial Mode.................................................................................................................... 605
User Subscriptions........................................................................................................ 605
Application Subscriptions............................................................................................... 606
Customer Operations Manager (COM)............................................................................ 607
Subscription Connection Operation................................................................................. 608
Subscription Network Requirements............................................................................... 609
Subscription Mode Ports................................................................................................ 610
Migrating Existing IP Office Systems to Subscription Mode............................................... 611
Chapter 52: General System Configuration................................................................. 612
Centralized System Directory......................................................................................... 612

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 Advice of Charge........................................................................................................... 616
Using Locations............................................................................................................ 617
Caller Display................................................................................................................ 617
Parking Calls................................................................................................................. 618
Automatic Intercom Calls............................................................................................... 619
Wide Band Audio Support.............................................................................................. 620
Media Connection Preservation...................................................................................... 621
Configuring IP Routes.................................................................................................... 622
Creating a Virtual WAN Port........................................................................................... 624
Chapter 53: On-boarding................................................................................................ 625
Configuring an SSL VPN using an on-boarding file........................................................... 625
Chapter 54: Fax Support................................................................................................ 627
Server Edition T38 Fax Support...................................................................................... 628
Chapter 55: Paging......................................................................................................... 630
Paging Capacity............................................................................................................ 630
Phone to Phone Paging................................................................................................. 631
Paging to an External Paging Device.............................................................................. 631
Mixed Paging................................................................................................................ 632
Paging Via Voicemail Pro............................................................................................... 633
Chapter 56: System Events............................................................................................ 636
Configuring Alarm Destinations...................................................................................... 637
Chapter 57: Certificate Management............................................................................. 638
Certificate Overview...................................................................................................... 638
Windows Certificate Store........................................................................................ 640
Certificate Support......................................................................................................... 643
Certificate File Naming and Format........................................................................... 643
Identity Certificate.................................................................................................... 644
Trusted Certificate Store.......................................................................................... 646
Signing Certificate................................................................................................... 647
Certificate File Import............................................................................................... 649
Chapter 58: Configuration for Emergency Calls.......................................................... 652
Emergency Call Indication.............................................................................................. 653
System Alarm Output..................................................................................................... 654
Chapter 59: Ring Tones.................................................................................................. 655
Chapter 60: Music On Hold............................................................................................ 657
System Source.............................................................................................................. 659
Alternate Source........................................................................................................... 659
Chapter 61: System Date and Time............................................................................... 663
System Date and Time Options...................................................................................... 663
Applying Daylight Saving............................................................................................... 664
Checking Automatic Time and Date Operation................................................................. 665
Manually Changing the System Date and Time................................................................ 666

June 2024 Administering Avaya IP Office using Manager 14

Comments on this document?
 Chapter 62: Configuring Time Profiles......................................................................... 667
Overriding a Time Profile................................................................................................ 668
Chapter 63: Applying Licenses..................................................................................... 670
PLDS licensing.............................................................................................................. 670
Web License Manager (WebLM)..................................................................................... 671
Server Edition Centralized Licensing............................................................................... 672
Distributing Server Edition Licenses................................................................................ 672
Nodal license distribution......................................................................................... 675
Centralized license distribution................................................................................. 676
Procedures for Applying Licensing.................................................................................. 677
Obtaining the Host ID of the WebLM Server.............................................................. 678
Installing a License File on the WebLM Server........................................................... 678
Configuring the Server Edition License Source.......................................................... 679
Uploading a PLDS License File to IP Office............................................................... 679
Configuring Server Edition Nodal Licensing............................................................... 680
Configuring Server Edition Centralized Licensing....................................................... 680
Configuring the License Server in an Enterprise Branch Deployment........................... 682
Converting from Nodal to Centralized Licensing............................................................... 682
Migrating Licenses to PLDS........................................................................................... 684
Chapter 64: Working with Templates............................................................................ 686
Saving Template files..................................................................................................... 686
Importing Trunk Templates............................................................................................. 687
Creating a Template in Manager..................................................................................... 687
Creating a New Record from a Template in Manager........................................................ 688
Creating an Analog Trunk Template in Manager............................................................... 689
Creating a New Analog Trunk from a Template in Manager............................................... 689
Applying a Template to an Analog Trunk.......................................................................... 689
Chapter 65: Configuring ARS........................................................................................ 691
Example ARS Operation................................................................................................ 692
ARS Operation.............................................................................................................. 693
ARS Short Codes.......................................................................................................... 695
Simple Alternate Line Example....................................................................................... 696
Simple Call Barring........................................................................................................ 697
User Priority Escalation.................................................................................................. 697
Time Based Routing...................................................................................................... 699
Account Code Restriction............................................................................................... 700
Tiered ARS Forms......................................................................................................... 700
Planning ARS............................................................................................................... 702
Chapter 66: Call Barring................................................................................................. 703
Applying Call Barring..................................................................................................... 703
Overriding call barring.................................................................................................... 704
Chapter 66: Configuring authorization codes.............................................................. 705

June 2024 Administering Avaya IP Office using Manager 15

Comments on this document?
 Entering an Authorization Code...................................................................................... 706
Chapter 66: Preventing Toll Bypass.............................................................................. 707
Configuring unknown locations....................................................................................... 708
Chapter 66: Configuring Call Admission Control........................................................ 709
Manager location tab..................................................................................................... 709
Assigning a network entity to a location...................................................................... 710
System actions at maximum call threshold...................................................................... 710
Example....................................................................................................................... 711
Chapter 67: Configure User Settings............................................................................ 713
User Management Overview.......................................................................................... 713
Configuring Gmail Integration......................................................................................... 715
Call Intrusion................................................................................................................. 716
Call Tagging.................................................................................................................. 719
Call Waiting.................................................................................................................. 719
Call Barring................................................................................................................... 720
Centralized Call Log...................................................................................................... 721
Centralized Personal Directory....................................................................................... 721
Account Code Configuration........................................................................................... 722
Setting a User to Forced Account Code..................................................................... 723
Malicious Call Tracing (MCID)........................................................................................ 724
Twinning....................................................................................................................... 724
Private Calls................................................................................................................. 727
System Phone Features................................................................................................. 728
The 'No User' User........................................................................................................ 729
Suppressing the NoCallerId alarm............................................................................ 730
Chapter 68: Avaya cloud authorization........................................................................ 731
Apple push notification services...................................................................................... 731
Enabling Apple push notifications............................................................................. 732
Chapter 69: Managing Users with LDAP...................................................................... 734
Performing LDAP Synchronization.................................................................................. 734
Creating a User Provisioning Rule for LDAP Synchronization............................................ 735
Chapter 70: Message Waiting Indication...................................................................... 737
Message Waiting Indication for Analog Phones................................................................ 737
Message Waiting Indication for Analog Trunks................................................................. 738
Chapter 71: Configuring User Rights............................................................................ 740
Adding User Rights........................................................................................................ 742
Creating a User Right Based on an Existing User............................................................ 742
Associating User Rights to a User................................................................................... 743
Copy User Rights Settings over a User's Settings............................................................ 743
Chapter 72: DND, Follow Me and Forwarding.............................................................. 744
Do Not Disturb (DND).................................................................................................... 745
Follow Me..................................................................................................................... 747

June 2024 Administering Avaya IP Office using Manager 16

Comments on this document?
 Forward Unconditional................................................................................................... 749
Forward on Busy........................................................................................................... 751
Forward on No Answer.................................................................................................. 753
Determining a User's Busy Status................................................................................... 755
Chaining....................................................................................................................... 756
Chapter 73: Hot Desking................................................................................................ 758
Hot Desking Operation................................................................................................... 759
Logging Out.................................................................................................................. 759
Hot Desking Controls..................................................................................................... 760
Hot Desking in an IP Office Network............................................................................... 760
Call Center Agents........................................................................................................ 761
Hot Desking Examples................................................................................................... 761
Scenario 1: Occasional Hot Desking......................................................................... 762
Scenario 2: Regular Hot Desking.............................................................................. 762
Scenario 3: Full Hot Desking.................................................................................... 762
Scenario 4: Call Center Hot Desking......................................................................... 763
Automatic Log Out......................................................................................................... 763
Chapter 74: Group Operation........................................................................................ 765
Group Types................................................................................................................. 768
Call Presentation........................................................................................................... 769
Group Member Availability............................................................................................. 771
Example Hunt Group..................................................................................................... 773
CBC/CCC Agents and Hunt Groups................................................................................ 774
Coverage Groups.......................................................................................................... 775
Chapter 75: Mobile Call Control.................................................................................... 776
Mobile Direct Access (MDA)........................................................................................... 779
Mobile Callback............................................................................................................. 781
Chapter 76: Transferring calls....................................................................................... 782
Transferring call notes................................................................................................... 782
Transferring call notes................................................................................................... 783
Off-Switch Transfer Restrictions...................................................................................... 784
Context Sensitive Transfer............................................................................................. 785
Dial Tone Transfer......................................................................................................... 786
Handsfree Announced Transfers.................................................................................... 788
One Touch Transferring................................................................................................. 790
Centrex Transfer........................................................................................................... 791
Chapter 77: Simultaneous mode................................................................................... 793
Simultaneous Mode Devices.......................................................................................... 793
Simultaneous Mode Notes............................................................................................. 793
Moving Calls Between Simultaneous Devices.................................................................. 794
Chapter 78: User Source Numbers............................................................................... 795
Individual User Source Numbers.................................................................................... 795

June 2024 Administering Avaya IP Office using Manager 17

Comments on this document?
 NoUser Source Numbers............................................................................................... 797
Chapter 79: Configuring IP Office Server Edition System Settings........................... 804
Synchronizing Server Edition passwords in Web Manager................................................ 804
Creating a common administration account..................................................................... 805
Voicemail Pro Administration.......................................................................................... 806
Server Edition Resilience............................................................................................... 806
Synchronizing the Configurations.................................................................................... 806
Starting Web Control..................................................................................................... 807
Part 7: Small Community Networking...................................................................... 808
Chapter 80: Small Community Networking.................................................................. 809
Supported Small Community Network Layouts................................................................ 810
Telephone Features Supported Across Server Edition and SCN Networks......................... 811
SCN Voicemail Support................................................................................................. 813
Enabling Small Community Networking........................................................................... 813
Setup the VoIP Line from System A to System B........................................................ 814
Setup the VoIP Line from System B to System A........................................................ 815
Small Community Network Remote Hotdesking............................................................... 815
Small Community Network Fallback................................................................................ 816
SCN Short Code Programming....................................................................................... 817
Chapter 81: Small Community Network Management................................................. 819
Enabling SCN Discovery................................................................................................ 819
Creating a Common Admin Account............................................................................... 820
Loading a Small Community Network Configuration......................................................... 821
Editing a Small Community Network Configuration........................................................... 822
System Inventory.......................................................................................................... 823
Chapter 82: Using the Network Viewer......................................................................... 825
Arranging the View........................................................................................................ 826
Adding a Line Within the Small Community Network........................................................ 826
Repairing a Line Within the Small Community Network..................................................... 827
Adding a System to the Small Community Network.......................................................... 827
Removing a Small Community Network Line................................................................... 828
Removing a System...................................................................................................... 828
Start System Status....................................................................................................... 829
Start Voicemail Pro........................................................................................................ 829
Add a Background Image............................................................................................... 829
Part 8: SIP Trunks............................................................................................................ 830
Chapter 83: SIP Trunk Overview.................................................................................... 831
Configuring a SIP Trunk................................................................................................. 831
SIP Line Requirements.................................................................................................. 832
Chapter 84: SIP Headers and URIs................................................................................ 836
SIP URI Formats........................................................................................................... 836
Standard SIP Headers................................................................................................... 837

June 2024 Administering Avaya IP Office using Manager 18

Comments on this document?
 Setting the SIP URI Host................................................................................................ 837
Setting the SIP URI Content........................................................................................... 838
Selecting the SIP Header Format Used........................................................................... 840
Chapter 85: Outgoing SIP Call Routing........................................................................ 841
SIP Outgoing Call Routing............................................................................................. 841
Anonymous SIP Calls.................................................................................................... 842
SIP ARS Response Codes............................................................................................. 843
Typical outgoing call scenarios....................................................................................... 845
Chapter 86: Incoming SIP Call Routing........................................................................ 848
SIP Short Codes........................................................................................................... 848
SIP Incoming Call Routing............................................................................................. 849
SIP Prefix Operation...................................................................................................... 851
Media path connection................................................................................................... 851
SIP Caller Name and Number Display............................................................................. 852
Typical incoming call scenarios....................................................................................... 853
Chapter 87: SIP messaging............................................................................................ 857
Codec selection............................................................................................................ 857
SIP DTMF Transmission................................................................................................ 858
Fax over SIP................................................................................................................. 859
SIP Call Hold Scenarios................................................................................................. 859
SIP Call Transfers (Refer).............................................................................................. 861
Ringback Tone.............................................................................................................. 862
Hold Reminders............................................................................................................ 863
Chapter 88: SIP Line Appearances............................................................................... 864
SIP Line Appearance Incoming Call Routing.................................................................... 864
SIP Line Appearance Outgoing Call Routing.................................................................... 864
SIP Line Appearance User Button Programming.............................................................. 865
Chapter 89: SIP Calling Number Verification (STIR/SHAKEN).................................... 866
The STIR/SHAKEN SIP Protocols.................................................................................. 867
Obtaining a call's number verification result..................................................................... 868
Setting the system's number verification default behavior................................................. 868
Enabling calling number verification on a SIP line............................................................ 869
SIP Calling Number Verification (STIR/SHAKEN)............................................................. 870
Changing the rejected call responses.............................................................................. 872
Changing the authentication header used........................................................................ 872
Customizing the call handling behavior........................................................................... 873
Call Records................................................................................................................. 873
Chapter 90: IP Office SIP trunk specifications............................................................. 875
SIP RFCs..................................................................................................................... 875
Transport protocols........................................................................................................ 877
Request methods.......................................................................................................... 877
Response methods........................................................................................................ 877

June 2024 Administering Avaya IP Office using Manager 19

Comments on this document?
 Headers....................................................................................................................... 878
Part 9: VMPro Auto Attendants................................................................................... 879
Chapter 91: Voicemail Pro Auto-Attendants................................................................ 880
Google TTS Prompt Language....................................................................................... 881
Text-to-Speech (TTS) Prompts....................................................................................... 881
Enabling Google Speech and the Default Voice............................................................... 882
Auto-Attendant Fallback Options.................................................................................... 883
Auto-Attendant Consent Example................................................................................... 883
Chapter 92: Voicemail Pro Auto-Attendant Settings................................................... 885
Auto-Attendant.............................................................................................................. 885
Actions......................................................................................................................... 889
Chapter 93: Voicemail Pro Auto-Attendant Actions.................................................... 892
Dial By Conference........................................................................................................ 892
Dial By Name................................................................................................................ 893
Dial By Number............................................................................................................. 895
Leave Message............................................................................................................. 896
Supervised Transfer...................................................................................................... 897
Park & Page................................................................................................................. 898
Replay Menu................................................................................................................. 900
Speak By Name............................................................................................................ 901
Speak By Number......................................................................................................... 902
Unsupervised Transfer................................................................................................... 903
Transfer to Auto Attendant............................................................................................. 904
Chapter 94: Recording Auto-Attendant Prompts (Voicemail Pro).............................. 906
Recording Auto-Attendant Prompts Using Short Codes.................................................... 906
Using Pre-Recorded Prompt Files................................................................................... 907
Recording Auto-Attendant Prompts Using Text-to-Speech................................................ 908
Recording User Name Prompts...................................................................................... 908
Chapter 95: Routing Calls to a Voicemail Pro Auto-Attendant................................... 910
Routing External Calls to an Auto-Attendant.................................................................... 910
Routing Internal Calls to an Auto-Attendant..................................................................... 910
Part 10: Conferencing.................................................................................................... 912
Chapter 96: Conferencing.............................................................................................. 913
Conference Types......................................................................................................... 913
Conference Participants................................................................................................. 914
User Conference Controls.............................................................................................. 914
Conference Capacities................................................................................................... 915
Conference ID Numbers................................................................................................ 916
Conference Notes......................................................................................................... 916
Conference Phones....................................................................................................... 917
Context Sensitive Conferencing...................................................................................... 918
Chapter 97: Ad-Hoc Conferencing................................................................................ 921

June 2024 Administering Avaya IP Office using Manager 20

Comments on this document?
 Dropping External Party Only Conferences...................................................................... 921
Adding Callers to an Ad-Hoc Conference........................................................................ 921
Chapter 98: Personal Meet-Me Conferences................................................................ 923
Setting a User’s Personal Conference PIN...................................................................... 923
Routing Internal Callers to a Meet-Me Conference........................................................... 924
Routing External Callers to a Meet Me Conference.......................................................... 924
Personal Meet-Me Conference Callflow........................................................................... 925
Chapter 99: System Conferences.................................................................................. 927
Adding a System Conference......................................................................................... 927
Editing a System Conference......................................................................................... 928
Deleting a System Conference....................................................................................... 928
System Conference Settings.......................................................................................... 929
Routing External Calls to a System Conference............................................................... 931
Part 11: Short Codes....................................................................................................... 932
Chapter 100: Short Code Overview............................................................................... 933
Short Code Characters.................................................................................................. 935
User Dialing.................................................................................................................. 940
Application Dialing......................................................................................................... 942
Secondary Dial Tone...................................................................................................... 942
? Short Codes............................................................................................................... 944
Short Code Matching Examples...................................................................................... 944
Default System Short Code List...................................................................................... 947
Chapter 101: Short Code Features................................................................................ 953
Auto Attendant.............................................................................................................. 956
Auto Intercom Deny Off.................................................................................................. 957
Auto Intercom Deny On................................................................................................. 957
Break Out..................................................................................................................... 958
Barred.......................................................................................................................... 958
Busy On Held................................................................................................................ 959
Call Intrude................................................................................................................... 960
Call Listen..................................................................................................................... 960
Call Park....................................................................................................................... 962
Call Park and Page........................................................................................................ 962
Call Pickup Any............................................................................................................. 963
Call Pickup Extn............................................................................................................ 963
Call Pickup Group......................................................................................................... 964
Call Pickup Line............................................................................................................ 965
Call Pickup Members..................................................................................................... 965
Call Pickup User............................................................................................................ 966
Call Queue................................................................................................................... 966
Call Record................................................................................................................... 967
Call Steal...................................................................................................................... 968

June 2024 Administering Avaya IP Office using Manager 21

Comments on this document?
 Call Waiting On............................................................................................................. 969
Call Waiting Off............................................................................................................. 969
Call Waiting Suspend..................................................................................................... 970
Cancel All Forwarding.................................................................................................... 970
Cancel Ring Back When Free........................................................................................ 971
Change Login Code....................................................................................................... 972
Clear After Call Work..................................................................................................... 972
Clear Call..................................................................................................................... 973
Clear CW...................................................................................................................... 973
Clear Hunt Group Night Service...................................................................................... 974
Clear Hunt Group Out Of Service.................................................................................... 975
Clear Quota.................................................................................................................. 975
Coaching Intrusion........................................................................................................ 976
Conference Add............................................................................................................ 976
Conference Meet Me..................................................................................................... 977
CW.............................................................................................................................. 978
Dial.............................................................................................................................. 979
Dial 3K1....................................................................................................................... 980
Dial 56K....................................................................................................................... 980
Dial 64K....................................................................................................................... 981
Dial CW........................................................................................................................ 981
Dial Direct..................................................................................................................... 982
Dial Direct Hot Line........................................................................................................ 982
Dial Emergency............................................................................................................. 983
Dial Extn....................................................................................................................... 983
Dial Fax........................................................................................................................ 984
Dial Inclusion................................................................................................................ 984
Dial Paging................................................................................................................... 985
Dial Physical Extension by Number................................................................................. 986
Dial Physical Extension By ID......................................................................................... 986
Dial Speech.................................................................................................................. 987
Dial V110...................................................................................................................... 987
Dial V120...................................................................................................................... 988
Dial Video..................................................................................................................... 988
Disable ARS Form......................................................................................................... 988
Disable Internal Forwards.............................................................................................. 989
Disable Internal Forward Unconditional........................................................................... 989
Disable Internal Forward Busy or No Answer................................................................... 990
Display Msg.................................................................................................................. 990
Do Not Disturb Exception Add........................................................................................ 991
Do Not Disturb Exception Delete.................................................................................... 992
Do Not Disturb On......................................................................................................... 993
Do Not Disturb Off......................................................................................................... 993

June 2024 Administering Avaya IP Office using Manager 22

Comments on this document?
 Enable ARS Form......................................................................................................... 994
Enable Internal Forwards............................................................................................... 994
Enable Internal Forward Unconditional............................................................................ 995
Enable Internal Forward Busy or No Answer.................................................................... 995
Extn Login.................................................................................................................... 995
Extn Logout.................................................................................................................. 997
Flash Hook................................................................................................................... 997
FNE Service................................................................................................................. 998
Follow Me Here............................................................................................................. 998
Follow Me Here Cancel.................................................................................................. 999
Follow Me To................................................................................................................. 999
Forward Hunt Group Calls On...................................................................................... 1000
Forward Hunt Group Calls Off...................................................................................... 1001
Forward Number......................................................................................................... 1001
Forward On Busy Number............................................................................................ 1002
Forward On Busy On................................................................................................... 1003
Forward On Busy Off................................................................................................... 1003
Forward On No Answer On.......................................................................................... 1004
Forward On No Answer Off.......................................................................................... 1004
Forward Unconditional On............................................................................................ 1005
Forward Unconditional Off............................................................................................ 1005
Group Listen Off.......................................................................................................... 1006
Group Listen On.......................................................................................................... 1006
Headset Toggle........................................................................................................... 1007
Hold Call..................................................................................................................... 1007
Hold CW..................................................................................................................... 1008
Hold Music.................................................................................................................. 1009
Hunt Group Disable..................................................................................................... 1009
Hunt Group Enable...................................................................................................... 1010
Last Number Redial..................................................................................................... 1010
MCID Activate............................................................................................................. 1011
Mobile Twinned Call Pickup.......................................................................................... 1011
Off Hook Station.......................................................................................................... 1012
Outgoing Call Bar Off................................................................................................... 1012
Outgoing Call Bar On................................................................................................... 1013
Private Call Off............................................................................................................ 1014
Private Call On............................................................................................................ 1014
Priority Call................................................................................................................. 1015
Record Message......................................................................................................... 1015
Relay On.................................................................................................................... 1016
Relay Off.................................................................................................................... 1017
Relay Pulse................................................................................................................ 1017
Resume Call............................................................................................................... 1018

June 2024 Administering Avaya IP Office using Manager 23

Comments on this document?
 Retrieve Call............................................................................................................... 1019
Ring Back When Free.................................................................................................. 1019
Secondary Dial Tone.................................................................................................... 1020
Set Absent Text........................................................................................................... 1020
Set Account Code....................................................................................................... 1022
Set Authorization Code................................................................................................ 1022
Set Fallback Twinning Off............................................................................................. 1023
Set Fallback Twinning On............................................................................................. 1023
Set Hunt Group Night Service...................................................................................... 1023
Set Hunt Group Out Of Service.................................................................................... 1024
Set Inside Call Seq...................................................................................................... 1025
Set Mobile Twinning Number........................................................................................ 1026
Set Mobile Twinning On............................................................................................... 1026
Set Mobile Twinning Off............................................................................................... 1026
Set No Answer Time.................................................................................................... 1027
Set Outside Call Seq................................................................................................... 1027
Set Ringback Seq........................................................................................................ 1028
Set Time Profile........................................................................................................... 1029
Set Wrap Up Time....................................................................................................... 1030
Speed Dial.................................................................................................................. 1031
Shutdown Embedded Voicemail................................................................................... 1031
Stamp Log.................................................................................................................. 1032
Startup Embedded Voicemail....................................................................................... 1032
Suspend Call.............................................................................................................. 1033
Suspend CW............................................................................................................... 1033
Start After Call Work.................................................................................................... 1034
Toggle Calls................................................................................................................ 1034
Unpark Call................................................................................................................. 1035
Voicemail Collect......................................................................................................... 1035
Voicemail Node........................................................................................................... 1037
Voicemail On............................................................................................................... 1037
Voicemail Off............................................................................................................... 1038
Voicemail Ringback On................................................................................................ 1038
Voicemail Ringback Off................................................................................................ 1039
Whisper Page............................................................................................................. 1040
Part 12: Button Programming................................................................................... 1041
Chapter 102: Button Programming Overview............................................................ 1042
Programming Buttons with IP Office Manager................................................................ 1043
Interactive Button Menus............................................................................................. 1044
Label Templates.......................................................................................................... 1044
Chapter 103: Button Programming Actions............................................................... 1046
Button Programming Actions Summary......................................................................... 1046
911-View.................................................................................................................... 1054

June 2024 Administering Avaya IP Office using Manager 24

Comments on this document?
 Abbreviated Dial.......................................................................................................... 1054
Abbreviated Dial Pause............................................................................................... 1055
Abbreviated Dial Program............................................................................................ 1055
Abbreviated Dial Stop.................................................................................................. 1056
Absent Message......................................................................................................... 1056
Account Code Entry..................................................................................................... 1056
ACD Agent Statistics................................................................................................... 1057
ACD Stroke Count....................................................................................................... 1057
Acquire Call................................................................................................................ 1058
AD Special Functions.................................................................................................. 1058
AD Special Function Mark............................................................................................ 1058
AD Special Function Wait............................................................................................. 1059
AD Suppress............................................................................................................... 1059
After Call Work............................................................................................................ 1060
Appearance................................................................................................................ 1061
Automatic Callback...................................................................................................... 1062
Auto-Intercom Deny..................................................................................................... 1063
Automatic Intercom...................................................................................................... 1064
Break Out................................................................................................................... 1064
Bridged Appearance.................................................................................................... 1065
Busy........................................................................................................................... 1066
Busy On Held.............................................................................................................. 1066
Call Forwarding All...................................................................................................... 1066
Call Intrude................................................................................................................. 1067
Call Listen................................................................................................................... 1068
Call Log...................................................................................................................... 1069
Call Park..................................................................................................................... 1069
Call Park and Page...................................................................................................... 1070
Call Park To Other Extension........................................................................................ 1071
Call Pickup................................................................................................................. 1072
Call Pickup Any........................................................................................................... 1072
Call Pickup Group....................................................................................................... 1073
Call Pickup Members................................................................................................... 1074
Call Queue................................................................................................................. 1074
Call Record................................................................................................................. 1075
Call Screening............................................................................................................ 1075
Call Steal.................................................................................................................... 1078
Call Waiting Off........................................................................................................... 1078
Call Waiting On........................................................................................................... 1079
Call Waiting Suspend................................................................................................... 1079
Cancel All Forwarding.................................................................................................. 1080
Cancel Leave Word Calling.......................................................................................... 1080
Cancel Ring Back When Free...................................................................................... 1081

June 2024 Administering Avaya IP Office using Manager 25

Comments on this document?
 Channel Monitor.......................................................................................................... 1081
Clear Call................................................................................................................... 1082
Clear CW.................................................................................................................... 1082
Clear Hunt Group Night Service.................................................................................... 1083
Clear Hunt Group Out Of Service.................................................................................. 1083
Clear Quota................................................................................................................ 1084
Coaching Intrusion...................................................................................................... 1084
Conference................................................................................................................. 1085
Conference Add.......................................................................................................... 1086
Conference Meet Me................................................................................................... 1086
Consult....................................................................................................................... 1088
Coverage Appearance................................................................................................. 1089
Dial............................................................................................................................ 1089
Dial 3K1..................................................................................................................... 1090
Dial 56K..................................................................................................................... 1090
Dial 64K..................................................................................................................... 1091
Dial CW...................................................................................................................... 1091
Dial Direct................................................................................................................... 1092
Dial Emergency........................................................................................................... 1093
Dial Inclusion.............................................................................................................. 1093
Dial Intercom............................................................................................................... 1094
Dial Paging................................................................................................................. 1095
Dial Physical Extn by Number...................................................................................... 1095
Dial Physical Number by ID.......................................................................................... 1096
Dial Speech................................................................................................................ 1096
Dial V110.................................................................................................................... 1097
Dial V120.................................................................................................................... 1097
Dial Video................................................................................................................... 1098
Directed Call Pickup.................................................................................................... 1098
Directory..................................................................................................................... 1099
Display Msg................................................................................................................ 1099
Do Not Disturb Exception Add...................................................................................... 1100
Do Not Disturb Exception Delete................................................................................... 1101
Do Not Disturb Off....................................................................................................... 1101
Do Not Disturb On....................................................................................................... 1102
Drop........................................................................................................................... 1102
Emergency View.......................................................................................................... 1103
Extn Login................................................................................................................... 1104
Extn Logout................................................................................................................. 1105
Flash Hook.................................................................................................................. 1105
Follow Me Here........................................................................................................... 1106
Follow Me Here Cancel................................................................................................ 1107
Follow Me To............................................................................................................... 1107

June 2024 Administering Avaya IP Office using Manager 26

Comments on this document?
 Forward Hunt Group Calls Off....................................................................................... 1108
Forward Hunt Group Calls On....................................................................................... 1108
Forward Number......................................................................................................... 1109
Forward On Busy Number............................................................................................ 1110
Forward On Busy Off.................................................................................................... 1111
Forward On Busy On.................................................................................................... 1111
Forward On No Answer Off........................................................................................... 1112
Forward On No Answer On........................................................................................... 1112
Forward Unconditional Off............................................................................................ 1113
Forward Unconditional On............................................................................................ 1114
Group......................................................................................................................... 1114
Group Listen On.......................................................................................................... 1115
Group Paging.............................................................................................................. 1116
Headset Toggle............................................................................................................ 1117
Hold Call..................................................................................................................... 1117
Hold CW..................................................................................................................... 1118
Hold Music.................................................................................................................. 1118
Hunt Group Enable...................................................................................................... 1119
Hunt Group Disable...................................................................................................... 1119
Inspect........................................................................................................................ 1120
Internal Auto-Answer................................................................................................... 1120
Last Number Redial..................................................................................................... 1121
Leave Word Calling..................................................................................................... 1121
Line Appearance......................................................................................................... 1122
MADN Call Appearance............................................................................................... 1123
Manual Exclude........................................................................................................... 1124
MCID Activate............................................................................................................. 1124
Monitor Analog Trunk MWI........................................................................................... 1125
Off Hook Station.......................................................................................................... 1125
Pause Recording......................................................................................................... 1126
Priority Call................................................................................................................. 1127
Priority Calling............................................................................................................. 1127
Private Call................................................................................................................. 1128
Relay Off..................................................................................................................... 1128
Relay On.................................................................................................................... 1129
Relay Pulse................................................................................................................. 1129
Resume Call............................................................................................................... 1130
Request Coaching Intrusion......................................................................................... 1131
Retrieve Call............................................................................................................... 1132
Ring Back When Free.................................................................................................. 1132
Ringer Off................................................................................................................... 1133
Self-Administer............................................................................................................ 1133
Send All Calls.............................................................................................................. 1135

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 Set Absent Text........................................................................................................... 1135
Set Account Code........................................................................................................ 1136
Set Hunt Group Night Service....................................................................................... 1137
Set Hunt Group Out Of Service..................................................................................... 1138
Set Inside Call Seq...................................................................................................... 1139
Set Night Service Destination....................................................................................... 1139
Set No Answer Time.................................................................................................... 1140
Set Out of Service Destination...................................................................................... 1140
Set Outside Call Seq.................................................................................................... 1141
Set Ringback Seq........................................................................................................ 1141
Set Wrap Up Time....................................................................................................... 1142
Speed Dial.................................................................................................................. 1142
Stamp Log.................................................................................................................. 1143
Stored Number View.................................................................................................... 1144
Suspend Call............................................................................................................... 1144
Suspend CW............................................................................................................... 1145
Swap CLID Name/Number........................................................................................... 1145
Time of Day................................................................................................................. 1145
Time Profile................................................................................................................. 1146
Timer.......................................................................................................................... 1148
Transfer...................................................................................................................... 1148
Toggle Calls................................................................................................................ 1149
Twinning..................................................................................................................... 1149
Unpark Call................................................................................................................. 1150
User........................................................................................................................... 1151
Visual Voice................................................................................................................ 1152
Voicemail Collect......................................................................................................... 1154
Voicemail Off............................................................................................................... 1155
Voicemail On............................................................................................................... 1155
Voicemail Ringback Off................................................................................................ 1156
Voicemail Ringback On................................................................................................ 1157
Whisper Page............................................................................................................. 1157
Part 13: Call Appearance Buttons............................................................................ 1159
Appearance Buttons.......................................................................................................... 1159
Chapter 104: Call Appearance Buttons....................................................................... 1161
Call Appearance Example 1......................................................................................... 1162
Call Appearance Example 2......................................................................................... 1162
How are Call Appearance Buttons Treated?................................................................... 1163
Call Appearance Button Indication................................................................................ 1164
Chapter 105: Bridged Appearance Buttons................................................................ 1166
Bridged Appearance Example 1.................................................................................... 1167
Bridged Appearance Example 2.................................................................................... 1167
Bridged Appearance Example 3.................................................................................... 1168

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 How are Bridged Appearances Treated?....................................................................... 1169
Bridged Appearance Button Indication........................................................................... 1170
Chapter 106: Call Coverage Buttons........................................................................... 1171
Call Coverage Example 1............................................................................................. 1171
Call Coverage Example 2............................................................................................. 1172
How is Call Coverage Treated?.................................................................................... 1173
Call Coverage Button Indication.................................................................................... 1174
Chapter 107: Line Appearance Buttons...................................................................... 1176
Line Appearance Example 1......................................................................................... 1177
Line Appearance Example 2......................................................................................... 1177
How are Line Appearances Treated?............................................................................ 1178
Line Appearance Button Indication................................................................................ 1179
Chapter 108: Appearance Button Features................................................................ 1181
Selected Button Indication............................................................................................ 1181
Idle Line Preference..................................................................................................... 1182
Ringing Line Preference............................................................................................... 1184
Answer Pre-Select....................................................................................................... 1186
Auto Hold.................................................................................................................... 1187
Ring Delay.................................................................................................................. 1188
Delayed Ring Preference............................................................................................. 1189
Collapsing Appearances.............................................................................................. 1191
Joining Calls................................................................................................................ 1192
Multiple Alerting Appearance Buttons............................................................................ 1194
Twinning..................................................................................................................... 1195
Busy on Held............................................................................................................... 1195
Reserving a Call Appearance Button............................................................................. 1196
Logging Off and Hot Desking........................................................................................ 1196
Applications................................................................................................................ 1197
Chapter 109: Programming Appearance Buttons...................................................... 1198
Appearance Function System Settings.......................................................................... 1200
Appearance Function User Settings.............................................................................. 1200
Programming Line Appearance ID Numbers.................................................................. 1202
Automatic Renumbering......................................................................................... 1202
Manual Renumbering............................................................................................ 1202
Outgoing Line Programming......................................................................................... 1203
Part 14: SMDR Call Records...................................................................................... 1205
Chapter 110: Appendix: SMDR Call Records............................................................. 1206
Enabling SMDR.......................................................................................................... 1206
SMDR Record Buffering............................................................................................... 1207
Checking SMDR Generation........................................................................................ 1207
SMDR Record Output.................................................................................................. 1207
SMDR Record Format................................................................................................. 1208

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 Call Times in SMDR.................................................................................................... 1208
SMDR Fields............................................................................................................... 1209
Chapter 111: SMDR Examples..................................................................................... 1214
SMDR Example: Lost Incoming Call.............................................................................. 1215
SMDR Example: Transfer............................................................................................ 1215
SMDR Example: Call Answered by Voicemail................................................................ 1216
SMDR Example: Call Transferred to Voicemail.............................................................. 1216
SMDR Example: Internal Call....................................................................................... 1216
SMDR Example: External Call...................................................................................... 1216
SMDR Example: Outgoing Call..................................................................................... 1217
SMDR Example: Voicemail Call.................................................................................... 1217
SMDR Example: Parked Call........................................................................................ 1217
SMDR Example: Incoming Call with Account Code........................................................ 1218
SMDR Example: Conference Using Conference Add Short Code.................................... 1218
SMDR Example: Conference Using Conference Button.................................................. 1219
SMDR Example: Adding a Party to a Conference........................................................... 1219
SMDR Example: Busy/Number Unavailable Tone.......................................................... 1220
SMDR Example: Call Pickup........................................................................................ 1220
SMDR Example: Internal Twinning................................................................................ 1220
SMDR Example: Park and Unpark................................................................................ 1221
SMDR Example: Distributed Hunt Group Call................................................................ 1221
SMDR Example: Voicemail Supervised Transfer............................................................ 1221
SMDR Example: Outgoing External Call........................................................................ 1222
SMDR Example: Rerouted External Call....................................................................... 1222
SMDR Example: External Forward Unconditional........................................................... 1222
SMDR Example: Call Transferred Manually................................................................... 1223
SMDR Example: Mobile Twinned Call Answered Internally............................................. 1223
SMDR Example: Mobile Twinned Call Answered at the Mobile Twin................................ 1224
SMDR Example: Mobile Twinned Call Picked Up Using the Twinning Button.................... 1224
SMDR Example: External Conference Party.................................................................. 1225
SMDR Example: Call Routed by Incoming Call Route.................................................... 1225
SMDR Example: Two Outgoing External Calls Transferred Together............................... 1225
SMDR Example: Authorization code............................................................................. 1226
SMDR Example: Internal Network Call.......................................................................... 1226
SMDR Example: Caller Consent Request...................................................................... 1226
Part 15: Miscellaneous................................................................................................. 1228
Chapter 112: Overview of Data Routing...................................................................... 1229
Network Address Translation (NAT).............................................................................. 1230
Dynamic Host Configuration Protocol (DHCP)............................................................... 1230
Simple ISDN Internet Connection................................................................................. 1231
ISDN Link Between IP Offices...................................................................................... 1231
Using a Dedicated T1/PRI ISP Link............................................................................... 1232
Tasks for Using a Dedicated T1/PRI ISP Link.......................................................... 1232

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 Create a New WAN Service................................................................................... 1232
Create the Virtual WAN Port................................................................................... 1234
Create an IP Route................................................................................................ 1234
T1 PRI Trunk........................................................................................................ 1235
Remote Access........................................................................................................... 1235
Creating a VoIP Link via the WAN Port Using PPP......................................................... 1239
Part 16: Further Help.................................................................................................... 1241
Chapter 113: Additional Help and Documentation.................................................... 1242
Additional Manuals and User Guides............................................................................ 1242
Getting Help................................................................................................................ 1242
Finding an Avaya Business Partner.............................................................................. 1243
Additional IP Office resources....................................................................................... 1243
Training...................................................................................................................... 1244

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Part 1: Introduction

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Chapter 1: Purpose

This document contains descriptions of the configuration fields and the configuration procedures
for administering Avaya IP Office Platform using the IP Office Manager application. This document
principally covers the IP Office Release 12.0.
Intended audience
The primary audience for the Administering Avaya IP Office using IP Office Manager is the customer
system administrators, implementation engineers and support and services personnel.
Related links
New in IP Office Release 12.0 on page 33

New in IP Office Release 12.0

The following changes apply for IP Office R12.0:
• Change of Linux Operating System
The version of Linux used by Linux-based IP Office servers has changed from CentOS to
Rocky Linux.
- The new Linux used is supported on 64-bit servers only.
- Booting from UEFI and BIOS is supported. Avaya recommends using UEFI where
possible.

Warning:
- For existing Linux-based IP Office systems upgrading to IP Office R12.0, you must
upgrade using the processes in Upgrading Linux-based IP Office Systems to R12.0.
• Display of Web Management Version
For Linux-based IP Office systems, the Control Unit details shown in IP Office Manager now
include details for the web management service.
• End of Support
The following are no longer supported:
- Web Collaboration
- Unified Communications Module (UCM)

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Purpose

The UCM uses a 32-bit processor and so is not supported by IP Office R12.0 and higher.
Existing systems using a UCM must migrate to an IP Office Application Server.
Related links
Purpose on page 33

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Chapter 2: IP Office Manager Overview

This documentation covers the use of the Avaya IP Office Manager. Manager runs on a Windows
PC and connects to the IP Office system via Ethernet LAN or WAN connections.
Important:
• IP Office Manager is an off-line editor. It receives a copy of the IP Office system's current
configuration settings. Changes are made to that copy and it is then sent back to the
system for those changes to become active. This means that changes to the active
configuration in the system that occur between IP Office Manager receiving and sending
back the copy may be overwritten. For example, this may affect changes made by a user
through their phone or voicemail mailbox after the copy of the configuration is received by
IP Office Manager.
Related links
IP Office Manager Modes on page 35
Security Configuration Mode on page 36
Standard Mode Configuration Mode on page 37
Server Edition Configuration Mode on page 39
Shell Server Mode on page 40
Backward Compatibility on page 41

IP Office Manager Modes

The menus and options displayed by Manager vary depending on the actions you are performing.
Manager runs in the following modes:

Mode Description
Basic Edition Mode This is the mode used when a Basic Edition configuration is opened. Basic
Mode includes systems running Partner, Norstar, or Quick Mode.
This is not covered in this document. Instead, refer to the separate IP Office
Basic Edition Manager manual.
Standard This is the mode used when a configuration from system running in Essential,
Configuration Mode Preferred, or Advanced Edition mode is loaded.
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IP Office Manager Overview

Mode Description
Server Edition This is the mode used when an IP Office Server Edition network configuration
Configuration Mode is opened.
Security Configuration Manager can be used to edit the security settings of IP Office systems.
Mode
Small Community Manager supports loading the combined configurations from systems in a
Network Management Small Community Network.
IP Office Shell Server The IP Office Shell Server is a single installation of selected IP Office
Mode applications running on Linux. You can use Manager to administer an IP Office
Shell Server.
Embedded File For systems with a memory card installed, Manager can be used to view and
Management manage the files stored on the card. Embedded File Management can be
accessed by selecting File | Advanced | Embedded File Management.
Upgrade Wizard The Upgrade Wizard is a component of Manager used to upgrade the firmware
run by the system.

Related links
IP Office Manager Overview on page 35

Security Configuration Mode

When Manager is in Security Mode, the screen elements shown are available.

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 Standard Mode Configuration Mode

Related links
IP Office Manager Overview on page 35

Standard Mode Configuration Mode

When Manager is in configuration mode, the screen elements shown are available. Some of these
elements can be customized, moved and hidden.

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IP Office Manager Overview

Manager Configuration Mode Screen Elements

Description
1. Title Bar
In addition to the application name, when configuration settings are loaded from a system, the title
bar displays the user name used to load the settings and the operator view applied. See Title Bar on
page 52.
2. Menu Bar
The options available with the drop down menus provided here change according to whether
Manager has a set of configuration or security settings loaded or not. See Menu Bar Commands on
page 75.
3. Main Toolbar
This toolbar provides icon shortcuts to the most frequently required configuration setting actions. See
Toolbars on page 52.
4. Navigation Pane
This pane shows icons for the different types of record that the configuration can contain. Each
type is followed by the number of records of that type already in the configuration. Selecting an
icon displays the matching records in the group pane and navigation toolbar. See The Navigation
Toolbar on page 54.
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 Server Edition Configuration Mode

Description
5. Group Pane
This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on column heading. Selecting a record in this pane displays
its details in the details pane. See The Group Pane on page 55.
6. Details Pane
This pane shows the configuration settings for a particular record within the configuration. The record
is selected using the navigation toolbar or using the navigation pane and group pane. See The
Details Pane on page 57.
7. Navigation Toolbar
This toolbar provides a set of drop downs which can be used to navigate to particular records in the
configuration settings. The selected options in the navigation pane, the group pane and the details
pane are synchronized with the navigation toolbar and vice versa. This toolbar is particularly useful if
you want to work with the group pane and or navigation pane hidden in order to maximize the display
space for the details pane. See The Navigation Toolbar on page 54.
8. Error Pane
This pane shows errors and warnings about the configuration settings. Selecting an item here loads
the corresponding record into the details pane. See The Error Pane on page 59.
9. Status Bar
This bar display messages about communications between Manager and systems. It also displays
the security level of the communications by the use of a padlock icon. See The Status Bar on
page 60.

Related links
IP Office Manager Overview on page 35

Server Edition Configuration Mode

When the configuration from a Server Edition solution is loaded into Manager, Manager switches
to Server Edition mode operation.

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IP Office Manager Overview

Related links
IP Office Manager Overview on page 35

Shell Server Mode

An IP Office shell server is an installation of selected IP Office applications running on an IP Office
Application Server. Since a shell server does not provide telephony, when you open a shell server
configuration, all telephony functions are disabled. The following functions are supported for a
shell server:
• Discovery
• Initial configuration utility.
• System status.
• Load, edit and save security settings.
• Load, edit, and save the configuration.

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 Backward Compatibility

• Erase configuration and security settings.

• Audit trail display.
• Web Control.
Related links
IP Office Manager Overview on page 35

Backward Compatibility
Manager is part of the IP Office Admin Suite of programs. The Manager application can be used
to manage configurations from systems running earlier software releases. Manager adjusts the
settings and fields that it shows to match the core software level of the system.
Manager is able display systems with software levels it does not support in the Select IP Office
discovery menu, however those systems are indicated as not supported.
Backwards compatibility is only supported for General Availability releases of IP Office software. It
is not supported for private builds.
Note that this document describes the current release. If you are running an earlier software
release, obtain the Manager document for the specific release from the Avaya support site.
Related links
IP Office Manager Overview on page 35

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Chapter 3: Getting Started

This section covers the installation of IP Office Manager and the initial loading of an IP Office
system configuration.
Related links
IP Office Manager PC requirements on page 42
Installing the IP Office Admin Suite on page 43
Downloading Manager Admin Lite on page 45
Starting Manager on page 45
Opening a Configuration on page 46
Login messages on page 48
Changing the Manager Language on page 50

IP Office Manager PC requirements

Supported Operating Systems
IP Office Manager is supported on Windows operating systems.
• IP Office Manager is only supported with the display font size set to 100%.
• If not already present, the required version of .NET Framework is installed as part of the IP
Office Manager installation.
• The System Status Application requires Java to also be installed on the PC. It is not installed
by the admin suite installer. This can be the run-time edition (JRE) or developers kit (JDK).
The application has been tested with Oracle and Azul Zulu versions of Java. The presence of
Java can be tested using the command java -version.

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 Installing the IP Office Admin Suite

Minimum PC Requirements
IP Office System RAM Available Minimum free Processor Network size
System (minimum or memory hard disk (similar or supported
higher) required for space higher)
Manager
operations
Standard Mode 4 GB 2 GB 6 GB Intel® Core™ i3 Not applicable.
or equivalent, 2
GHz minimum
Server Edition 4 GB (32 bit 2 GB 6 GB Intel® Core™ i3 Up to 32 nodes
OS) or equivalent, 2
GHz minimum
Server Edition 8 GB (64 bit 4 GB 6 GB Intel® Core™ i5 Up to 150
OS) or equivalent, 2 nodes
GHz minimum

Ports
For information on port usage, see refer to https://ipofficekb.avaya.com/businesspartner/ipoffice/
mergedProjects/general/port_matrix/index.htm.
Related links
Getting Started on page 42

Installing the IP Office Admin Suite

IP Office Manager is a component in a set of applications called the IP Office Admin suite. These
applications are used to configure and monitor an IP Office system.
• The installer for the IP Office Admin suite can be download from the Avaya support website
(https://support.avaya.com).
• Users of IP Office Web Manager can download a separate installer called Admin Lite. See
Downloading Manager Admin Lite on page 45.
In addition to installing IP Office Manager, the installers include options to install the following
additional applications:
Application Description
System Monitor This is a tool for system installers and maintainers. Interpreting the information
output by System Monitor requires detailed data and telecoms knowledge.
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Getting Started

Application Description
System Status This is a Java application that can be used to monitor the status of the system
Application such as extension, trunks and other resources. It displays current alarms and most
recent historical alarms.
• The System Status Application requires Java to also be installed on the PC. It is
not installed by the admin suite installer. This can be the run-time edition (JRE) or
developers kit (JDK). The application has been tested with Oracle and Azul Zulu
versions of Java. The presence of Java can be tested using the command java
-version.

Note:
This installation process installs the required version of Windows .NET if not already present.
This may require some systems to restart and the installation process to then be restarted.
Procedure
1. Depending on the version of installer:
• IP Office Admin Suite:
a. Unzip the downloaded installer file.
b. Locate and right-click on the setup.exe file. Select Run as Administrator.
• IP Office Admin Lite:
a. Right-click on the downloaded IPOAdminLite.exe file. Select Run as
Administrator.
2. Select the language you want to use for the installation process. This does not affect the
language used by Manager when it is run. Click Next >.
3. If an upgrade menu appears, it indicates that a previous installation has been detected.
Select Yes to upgrade the existing installed applications.
4. If required select the destination to which the applications should be installed. We
recommend that you accept the default destination. Click Next >.
5. Select which applications in the suite should be installed. Click on the next to each
application to change its installation selection. When you have selected the installations
required, click Next >.
6. The applications selected are now ready to be installed. Click Next >.
7. Following installation, you are prompted whether you want to run Manager.
8. On some versions of Windows, you may be required to restart the PC. Allow this to happen
if required.
Related links
Getting Started on page 42

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 Downloading Manager Admin Lite

Downloading Manager Admin Lite

About this task
The Admin Lite installer installs the same set of applications as the full IP Office Admin suite but
with the limitations listed below. You can download the installer for Admin Lite from your own IP
Office system.
Feature Description
Help Files Only English language help file are included.
Phone Files Phone firmware and other files used to upgrade phones are not included in the admin lite
installation.
Memory Cards The files used to upgrade IP Office systems directly or through actions such as SD card
Folder recreation are not included.

Before you begin

Ensure you have IP Office administrator name and password.
Procedure
1. Start your browser and enter https://<IP Office system IP address>/
index.html. For example https://192.168.42.1/index.html.
2. From the list of links, click IP Office Web Manager.
3. Login using your administrator name and password.
4. Click Solution.
5. Click on the icon to the right of the server name and select Platform View.
6. In the new browser window, click on the AppCenter tab.
7. From the set of Download Applications, download the AdminLite installer.
Next steps
• See Installing the IP Office Admin Suite on page 43.
Related links
Getting Started on page 42

Starting Manager
No name or password is required to start Manager. A name and password is only required when
connecting with a system.

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Getting Started

When started, by default Manager will attempt to discover any systems on the network. If it finds
any it will display a list from which you can select the system required.
1. Select Start and then Programs or All Programs depending on the version of Windows.
Select the IP Office program group.

2. Select Manager. If a Windows Security Alert appears select Unblock to allow Manager
to run.
3. By default Manager will scan the network for any systems. What appears next depends on
whether it finds any systems.
• If Manager finds multiple systems, the Select IP Office window displays a list of those
systems from which you can select the one whose configuration you want to edit. If you
want to open a configuration go to Opening a Configuration. If you don't want to load a
configuration click on Cancel.
• If it finds a single system, it will attempt to open the configuration of that system by
displaying the Configuration Service User Login window..
• If no systems are found or you cancel the steps above, the Manager simplified view is
displayed. Use the simplified view to select one of the following action:
- Create an Offline Configuration
- Open a Configuration from a System
- Read a Configuration from a File
Related links
Getting Started on page 42

Opening a Configuration
The initial IP address ranges in which Manager searches for systems is set through the File |
Preferences | Discovery. By default, Manager scans the local network of the Manager PC.
1. Start Manager.
• If Manager is already started and a configuration is open in it, that configuration must be
closed first.
• If Manager is set to Auto Connect on start up, it will scan for systems automatically
and either display the list of systems discovered or automatically start login to the only
system discovered.
• Otherwise, select File | Open Configuration.
2. The Select IP Office window opens, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the
configuration of those systems cannot be opened using Manager in Advanced View

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 Opening a Configuration

mode and the configuration of a Primary Server can only be opened if the Open with
Server Edition Manager option is also selected.
• If Manager has been set with SCN Discovery enabled, systems in a Small Community
Network are grouped together. The checkbox next to the network name can be used
to load the configurations of all the configurations into Small Community Network
management mode.
• If the system required was not found, the Unit/Broadcast Address used for the search
can be changed. Either enter an address or use the drop-down to select a previously
used address. Then click Refreshto perform a new search.
• A list of known systems can be stored using Known System Discovery.
• Manager can be configured to search using DNS names.
• Systems found but not supported by the version of Manager being used will be listed as
Not Supported.

• If the system detected is running software other than from its primary folder, a
warning icon will be shown next to it. The configuration can still be opened but only as a
read-only file.
3. When you have located the system required, check the box next to the system and click
OK.
• If the system selected is a Server Edition system and Manager is not running in Server
Edition mode, an Open with Server Edition Manager checkbox is shown and pre-
selected. Clicking OK will switch Manager to its Server Edition mode before loading the
configuration.
4. The system name and password request is displayed. Enter the required details and click
OK. The name and password used must match a service user account configured within
the system's security settings.
5. Additional messages will inform you about the success or failure of opening the
configuration from the system. See Login messages on page 48.
6. The method of connection, secure or insecure, attempted by Manager is set by the
application's Secure Communications preferences setting.

• When Secure Communications is set to On, a padlock icon is displayed at all times
in the lower right Manager status field.
• New installations of Manager default to having Secure Communications enabled. This
means Manager by default attempts to use secure communications when opening a
configuration.
• For Server Edition systems, Manager always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager
offers to fallback to using unsecured communications.
7. Following a successful log in, the configuration is opened in Manager. The menus and
options displayed depend on the type of system configuration loaded.

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Getting Started

Related links
Getting Started on page 42

Login messages
While attempting to login to a system, various messages may be displayed.
Login History
When logging in, user information details about the last login attempt, with date and time are
displayed.
Security Banner
You can set up an IP Office Manager security banner to include custom text. For example:
• Informative messages: To indicate the server role in a network, this may be useful in a
network with multiple servers.
• Warning messages: To indicate a warning to restrict any system modification during the
upgrade or backup process.
• General purpose messages: To indicate unauthorized access or system security restrictions.
For example: This system is restricted solely to authorized users for legitimate business
purposes. The actual or attempted unauthorized access, use, or modification of this system is
strictly prohibited.
To set up a security login banner, before logging into the IP Office Manager, do the following:
1. Open a .txt file.
2. Enter the required custom text.
3. Save it as etcissue.txt in the IP Office Manager application's installation folder:
File Path
Pre-R11.1 FP2 SP3 C:\Program Files (x86)\Avaya\IP Office\Manager
Full Admin Suite C:\Program Files (x86)\Avaya\IP Office Admin
Suite\Manager
Admin Lite C:\Program Files (x86)\Avaya\IP Office Lite\Manager

Configuration Not Loaded Messages

Message Description
Access Denied Displayed as the cause if the service user name/password were incorrect, or the service
user has insufficient rights to read the configuration. The Retry option can be used to log
in again but multiple rejections in a 10 minute period may trigger events, such as locking
the user account, set by the Password Reject Limit and Password Reject Action options
in the systems security settings.
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 Login messages

Message Description
Failed to Displayed as the cause if the network link fails, or the secure communication mode is
communicate with incorrect (for example Manager is set to unsecured, but the system is set to secure only).
system
Account Locked The account of the service user name and password being used is locked. This can
be caused by a number of actions, for example too many incorrect password attempts,
passing a fixed expiry date, etc. The account lock may be temporary (10 minutes)
or permanent until manually unlocked. An account can be enabled again through the
system's security settings.

Additional Messages
Message Description
Your service user Indicates that an Account Expiry date has been set on the system service user account
account will expire and that date is approaching. Someone with access to the system's security settings will
in X days be required unlock the account and set a new expiry date.
Your password will Indicates that password aging has been configured in the system's security settings. If
expire in X days. your password expires, someone with access to the system's security settings will be
Do you wish to required to unlock the account.
change it now?
Limit of concurrent Indicates that the administrator account has been used for more than five concurrent
sessions per user sessions. IP Office allows five concurrent sessions using one administrator account.
exceeded If five sessions are already on, logging in for the sixth session fails and Web
Manager displays an error message Limit of concurrent sessions per user
exceeded. Note that the following are also considered as a session:
• If Manager is connected with IP Office Server Edition through SE Central Access.
• If the same administrator account is used to log in to any of the IP Office third party
applications developed using the Management SDK client.
Change password Through the system's security settings, a service user account can be required to change
their password when logging in. The menu provides fields for entering the old password
and new password.
Contact This message displays if a Manager user with administrator rights has entered their
Information Check contact information into the configuration. For example to indicate that they do not
- This want the configuration altered while a possible problem is being diagnosed. The options
configuration is available are:
under special
• Cancel - Select this option to close the configuration without making any changes.
control
• Set configuration alteration flag - Select this option if the configuration is being
opened because some urgent maintenance action. When the configuration is next
opened, the fact that it has been altered will be indicated on the System > System tab.
• Delete Contact Information - Select this option to take the system out of special
control.
• Leave contact information and flags unchanged (Administrators only) - This
option is only available to service users logging in with administrator rights.

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Getting Started

Related links
Getting Started on page 42

Changing the Manager Language

About this task
The Manager application can run in multiple languages. By default it tries to use the best match to
the PC's regional location settings, otherwise it uses UK English.
The process below can be used to run Manager in one of its supported languages. Some
components in the application such as the error pane, tooltips, and pop-up messages continue
to use the System Locale language.
Note:
The IPOAdminLite does not support other languages. Only English language Manager.chm
help file is present in IPOAdminLite.exe installation.
Procedure
1. Create a Windows shortcut to the Manager application .exe file. By default this file is
located in the following folder:
File Path
Pre-R11.1 FP2 SP3 C:\Program Files (x86)\Avaya\IP Office\Manager
Full Admin Suite C:\Program Files (x86)\Avaya\IP Office Admin
Suite\Manager
Admin Lite C:\Program Files (x86)\Avaya\IP Office Lite\Manager

2. Right-click on the shortcut and select Properties.

3. The Target field can be used to specify the locale setting that Manager should use.
For example, for Italian the Target should have -locale:it-IT added
to the end. For example: "C:\Program Files (x86)\Avaya\IP
Office\Manager\Manager.exe" -locale:it-IT.

Table 1: Manager Locales

Manager Language Shortcut Locale Setting

Brazilian Portuguese -locale:pt-Br
Chinese (Simplified) -locale:zh-Hans
Dutch -locale:nl-NL
French -locale:fr-FR
German -locale:de-DE
Table continues…

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 Changing the Manager Language

Manager Language Shortcut Locale Setting

Italian -locale:it-IT
Mexican Spanish -locale:es-MX
Russian -locale:ru-RU
US English -locale:en-US

4. Click OK.
5. Manager will now run in the selected language when launched using the updated shortcut.
Related links
Getting Started on page 42

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Chapter 4: Manager User Interface

This section of the documentation covers the operation of Manager when being used to edit the
configuration of a system running in Standard Mode. Much of it is also applicable for when also
editing the configuration of systems running in Server Edition mode. Additional Server Edition Mode
functions are detailed in the next chapter.
Related links
Title Bar on page 52
Toolbars on page 52
The Navigation Pane on page 54
The Group Pane on page 55
The Details Pane on page 57
The Error Pane on page 59
The Status Bar on page 60

Title Bar
The Manager title bar shows the following information.
• The Manager application version.
• The system name of the system from which the currently loaded configuration was received.
• The software level of the system's control unit.
• The service user name used to receive the configuration and that user's associated operator
rights.
Related links
Manager User Interface on page 52

Toolbars
Manager displays the following toolbars:
• Main Toolbar

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 Toolbars

• Navigation Toolbar
• Details Toolbar
Related links
Manager User Interface on page 52

The Main Toolbar

The Main toolbar is displayed at the top of the Manager window, just below the menu bar. This
toolbar is also available when Manager is in security mode. However many of the controls will not
function in security mode.

Open Configuration from a System Advertises to the address currently shown in the Manager's
title bar for any available systems. A list of responding systems is then displayed. When a system
is selected from this list, a valid user name and password must be entered. Equivalent to File |
Open Configuration.

Open Configuration File Open a configuration file stored on a PC. The button can be clicked to
display a browse window. Alternatively the adjacent arrow can be used to drop-down a list of the
last 4 previously opened configuration files. Equivalent to File | Offline | Open File.

Save Configuration File The action of this icon depends on whether the currently loaded
configuration settings were received from a system or opened from a file stored on PC. If the
former applies, the menu sending the configuration back to the system is displayed. In the latter
case, the file changes are saved to the original file. Equivalent to File | Save Configuration.

Collapse All Groups Causes all symbols in the navigation pane to be collapsed to symbols.

Show/Hide the Navigation Pane

Show/Hide the Group Pane

Show/Hide the Error Pane

Validate Configuration Runs a validation on all the currently loaded configuration settings.
The results appear in the error pane. By default the configuration is automatically validated
when loaded and changes are validated when made, however the validation preferences can
be changed through File | Preferences | Validation.

Create New Configuration Runs a series of dialogs that create a new configuration from
scratch.

Connect To For a standalone system, start the process of adding it to a multi-site network. Not
available in Server Edition mode.

Voicemail Pro Client Launch the Voicemail Pro client if also installed on the Manager PC.

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Manager User Interface

Server Edition Solution View Switch to the solution view. This option is only shown when
Manager is running in Server Edition mode.

The Navigation Toolbar

This toolbar provides drop down lists which can be used to navigate to particular records in the
configuration settings. The selected options in the navigation pane, group pane and the details
pane are synchronized with the navigation toolbar and vice versa. This toolbar is particularly
useful if you want to work with the group pane and or navigation pane hidden in order to maximize
the display space for the details pane.

This toolbar is not available when Manager is in security mode.

The Details Toolbar

This toolbar is shown in the top-right of the details pane. The options within the toolbar may vary
or be greyed out depending on the actions allowed for a particular configuration record.

Create a New Record The arrow is used to select the record type to be created. For
example; when adding an extension clicking may allow selection of a VoIP Extension or IP
DECT Extension.

Export as Template Save the current record as a tempate. The template can then be used to
create new records.
Delete Current Record Delete the currently displayed record.
Validate Current Record By default records are validated when opened and when edited. This
is set through the Manager application's validation settings.
< > Previous Record/Next Record Click < or > at the top-right to move to the previous or next
record.

The Navigation Pane

This pane shows icons for the different types of record that the configuration can contain. Each
type is followed by the number of records of that type already in the configuration. When Manager
is used in security mode, this pane is also used by Manager in security mode to display records
for security settings.
Selecting an icon displays the matching records in the group pane, navigation toolbar and details
pane. Note that Manager is used to configure different types of system. Therefore the icons shown

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 The Group Pane

may vary depending on the type of system you are configuring. For descriptions of the different
icons refer to Configuration Settings.
The information in the pane also depends on whether the group pane is visible or not. If the group
pane is visible, the navigation pane just shows icons for accessing which types of records should
be shown in the group pane. The group pane can then be used to select which of those records
is currently shown in the details pane. If the group pane is not visible, the navigation pane shows
icons for each type of records and under those icons for each individual record. The navigation
pane can then be used to select which of those records is currently shown in the details pane.
Related links
Manager User Interface on page 52

Expanding and Collapsing the Navigation Tree

Where or icons appear in the pane, they allow the structure to be expanded or collapsed.
When the group pane is hidden, and icons are shown for each record type and allow the
record type to be expanded to display all the existing records of that type.

The icon in the main toolbar can also be used to collapse all the expanded record types shown
in the navigation pane.

The Group Pane

This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on a column heading. Selecting a record in this pane
displays its details in the details pane.
The icons used in the pane may vary according to the state of the record. For example, some of
the users shown in this example have been configured for hot desking. This pane is also used by
Manager in security mode to display records for security settings.
Related links
Manager User Interface on page 52

Sorting the List

About this task
The records shown in the group pane can be sorted using any of the columns displayed.
Procedure
1. To sort the list using the details in a particular column, click on the column header.
2. Clicking on the same column header again reverses the sort order.

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Manager User Interface

Customizing the Columns Displayed

About this task
For each record type, which details are shown in the group pane can be customized. Also the
order of the column can be adjusted.
Procedure
1. Right-click on the pane and select Customize Columns.
2. To add a column, selects its name in the left-hand Available Columns list and click >>to
move it to the right-hand Selected Columns list.
3. To remove a column, select its name in the right-hand Selected Columns list and click <<
to move it to the left-hand Available Columns list.
4. To change the order of the Selected Columns, click on a column name and use the ^ and
V controls.
5. Click OK.

Changing the Column Widths

Procedure
1. In the column headers, place the cursor over the border between two columns.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold
the cursor.
3. Drag the border to the required position and release the cursor.

Adding a New Record

About this task
The group pane can be used to add a new record of the type currently displayed.
Right-click on the pane and select New.
A arrow symbol next to New indicates that you can select a particular type of new record to
create. Click the arrow and select an option from the list.
Procedure
1. Use the details pane to configure the new record.
2. Click OKin the details pane.

Deleting an Record
Procedure
1. Select the record to be deleted by clicking on it.
2. Right-click on the pane and select Delete.

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 The Details Pane

Validating an Record
Procedure
1. Select the record to be validated by clicking on it.
2. Right-click on the pane and select Validate.

Show in Groups
About this task
This command groups the items shown in the group pane. The grouping method will vary
depending on the record type being listed. For example, short codes are grouped based on short
code feature type such as all forwarding short codes together.
Procedure
Right-click on the pane and select Show In Groups.

The Details Pane

Whenever a selection is made through the group pane or the navigation toolbar, the settings for
the matching record are shown in the details pane. This pane is also used by Manager in security
mode to display records for security settings.
The details are grouped into tabs. The tabs available may vary depending on what particular type
of record is being viewed.
Individual settings may also be grayed out. This indicates that they are either for information only
or that they cannot be used until another setting is enabled.
The top-left icon indicates the following:
Locked
Indicates that you can view the settings but cannot change them.
Editable
Indicates that you can change the settings if required.
Changed
Indicates that the settings have been changed since the tab was opened. Click OK to save
the changes or Cancel to undo.

Various icons may appear adjacent to settings:

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Manager User Interface

Locked Setting
The setting cannot be changed through this tab. This icon appears on user settings where the
user is associated with User Rights that controls the setting.
Information
Indicates a value which does not have to be set but may be useful if set.
Warning
A warning indicates a configuration setting value that is not typical and may indicate
misconfiguration.
Error - An error indicates a configuration setting value that is not supported by the system. Such
settings may cause the system to not operate as expected.

Related links
Manager User Interface on page 52
Managing Records on page 58

Managing Records
Procedure
1. Edit a record
a. The method of entering a record varies as different fields may use different methods.
For example text record boxes or drop down lists.
b. By default when changes are made, they are validated once another field is selected.
See File | Preferences | Validation.
c. Click on OK at the base of the details pane to accept the changes or click on Cancel
to undo the changes.
2. Add a record.
a. Click at the top-right of the details pane.
b. Select the type of record required. For example, with extensions you can select from
H.323 Extension or SIP Extension.
3. Delete a record.
Click at the top-right of the details pane.
4. Validate a record.
Click at the top-right of the details pane.
5. Move to the previous or next record.
Click <or > at the top-right to move to the previous or next record.
6. Select a new tab.
a. To view the detail stored on a particular tab, click on the name of that tab.

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 The Error Pane

b. If the tab required is not shown, use the controls if shown on the right to scroll
through the available tabs. The tabs available may vary depending on what particular
type of record is being viewed.
Related links
The Details Pane on page 57

The Error Pane

Validation is a process where Manager checks configuration records for errors or for values for
which it regards as requiring a warning. The results of this checking are shown by icons next to
the field that caused the error or warning, All errors and warnings are also listed in the Error Pane.
By default validation is performed automatically whenever a configuration file is opened and when
any field is edited. However, if required, the use of automatic validation can be controlled through
the settings on the File > Preferences > Validation tab.
The icons used for errors and warnings are as follows. These are shown in the error pane
and also next to the related field in the details pane. In the details pane, the error or warning
description is shown when the cursor is hovered over the icon.
Icon Description
Error
An error indicates a configuration setting value that is not supported by the system. Such settings
are likely to cause the system to not operate as expected.
Warning
A warning indicates a configuration setting value that is not typical and may indicate
misconfiguration.
Information
Typically indicates a setting which may be useful to set.

Related links
Manager User Interface on page 52

Altering the Automatic Validation Settings

About this task
The settings for automatic validation are adjustable.
Procedure
1. Select File | Preferences.
2. Select the Validation tab.

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Manager User Interface

Select the options required.

• Validate configuration on open Automatically validate configuration files when they
are opened in Manager.
• Validate configuration on edit Validate the whole configuration when OK is clicked
after editing a record. For large configurations, disabling this option removes the delay
caused by validating the configuration after every edit.
• Prompt for configuration validation on save or send If selected, when saving or
sending a configuration, a prompt is displayed asking whether the configuration should
be validated. If validation is selected and error are found, the send or save process is
canceled. This option is disabled if Validate configuration on edit is selected.
3. Click OK.

Revalidating Configuration Settings

About this task
If necessary, you can force a validation check of the whole configuration or of the current record
shown in the details pane.
Procedure
1. To validate the whole configuration, click in the main toolbar.
2. For a particular record, click in the details pane.

Viewing an Error or Warning

Procedure
1. Clicking on an error or warning in the error pane will load the matching record tab into the
details pane.
2. The < and > can be used to move to the next error or warning in the error pane.

The Status Bar

The status bar at the base of the Manager screen is used to display icons and messages about
communications between Manager and systems. If the Manager is also acting as a BOOTP and
TFTP server it will also show BOOTP and TFTP messages.
A padlock icon is displayed whenever the Manager communications settings are set to secure.
This indicates all attempted configuration and security settings exchanged will be attempted over a
secure TLS link:
Status bar messages display information about communications the Manager application receives.
Some typical status bar messages are listed below.
Ready

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 Configuring the Interface

This message is normally seen when Manager has just started and no configuration has been
received.
Received BOOTP request for 001125465ab2, unable to process
Manager is acting as a BOOTP server. It has received a BOOTP request that does not match a
system listed in its BOOTP records. The cause may be a device or application, other than an IP
Office, that also uses BOOTP.
TFTP: Received TFTP Error "NotFound" from 192.168.42.1
An attempt to receive settings from or send settings to the system failed. The most probable cause
is a name or password error.
TFTP: Received 17408 bytes for Marks_Test
Manager has received configuration settings from the named system using TFTP.
Sent 100% of C:\Program Files\Avaya\IP Office\Manager\b10d01b2_3.bin
Manager has sent the indicated file in response to a BOOTP request.
Related links
Manager User Interface on page 52

Configuring the Interface

The Manager configuration settings interface can be customized in a number of ways. These
changes are remembered the next time Manager is started.
Related links
Manager User Interface on page 52

Resizing the Manager Window

About this task
When the Manager window is not maximized or minimized, it size can be adjusted.
Procedure
1. Place the cursor over the edge of the current window.
2. When the cursor changes to a double-headed arrow, click and hold the cursor.
3. Drag the edge to the required position and then release the cursor.

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Manager User Interface

Moving the Border Between the Panes

About this task
The border between the visible panes can be adjusted. Note that this is a proportional rather than
exact position. If the whole window size is altered, the border position may also move.
Procedure
1. Place the cursor over the border between two panes.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold
the cursor.
3. Drag the border to the required position and release the cursor.

Showing or Hiding Toolbars

About this task
The different toolbars can be hidden if not required.
Procedure
1. Select View and then Toolbars.
Those toolbars currently shown are indicated by a tick mark.
2. To show or hide a toolbar, click on its name.

Moving Toolbars
About this task
The position of the Manager toolbars can be moved. Note that when moving a toolbar, the other
toolbars and panes may adjust their size or position to ensure that all the toolbar icons remain
visible.
Procedure
1. Place the cursor over the end of the toolbar.
2. When the cursor changes to a four-way arrow, click and hold the cursor.
3. Move the toolbar to the required position and release the cursor.

Showing or Hiding Panes

About this task
The details pane cannot be hidden. The navigation pane, group pane and error pane can be
shown or hidden. To do this use either of the following methods.
From the main toolbar, use the following icons:

• Hide/Show Navigation Pane.

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 Configuring the Interface

• Hide/Show Group Pane.

• Hide/Show Error Pane.

or
Procedure
1. Select View.
Those panes currently shown are indicated by a tick mark.
2. To show or hide a pane, click on its name.

Changing the Position of the Details Pane

About this task
When the group pane is visible, the details pane is shown either below it or to its right. This
position can be adjusted.
Procedure
1. Select View and then Details Pane.
2. The current position setting is indicated by a tick mark.
3. To select a position, click on it.

Changing the Size of Configuration Icons

About this task
The size of the icons used on the navigation pane and details pane can be adjusted.
Procedure
1. Select File and then Preferences.
2. Select the Visual Preferences tab.
3. Select the required icon size from Small, Medium or Large.
4. Click OK.

Changing Tab Display

About this task
For records with more than two tabs, you can select whether Manager should use controls or
arrange the tabs as multiple rows when necessary.
Procedure
1. Select Files | Preferences | Visual Preferences.
2. Select Multiline Tabs.

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Manager User Interface

3. Click OK.

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Chapter 5: Working with the Server Edition
Manager User Interface

When using IP Office Manager to manage a Linux-based network of IP Office systems such as
Server Edition, the IP Office Manager interface supports a number of additional features.
Related links
Server Edition Solution View on page 65
System Inventories on page 69
Default Settings on page 69
Record Consolidation on page 70
Telephone Features Supported Across Server Edition and SCN Networks on page 71

Server Edition Solution View

When the configuration of a Server Edition solution is loaded into Manager, Manager starts with
the Server Edition Solution View menu. This menu includes the system inventory of the servers,
links for launching various functions and a summary table of the servers and the links between the
servers.
Related links
Working with the Server Edition Manager User Interface on page 65
Displaying the Server Edition Solution View on page 66
The Solution View Network Display on page 66
Opening additional applications on page 67
Setting all nodes to Subscription mode on page 68
Setting all nodes to Select mode on page 68
Setting the License Source in Server Edition View on page 68
Adding a Server in the Server Edition Solution View on page 69

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Working with the Server Edition Manager User Interface

Displaying the Server Edition Solution View

Manager normally starts with the Server Edition Solution View when the configuration for a Server
Edition network is loaded. However, if required, to return to the solution view do one of the
following.

• Click on the Server Edition Solution View icon in the toolbar.

• Click on the Solution icon in the navigation pane.
Related links
Server Edition Solution View on page 65

The Solution View Network Display

The table at the bottom of the solution view give a quick overview of all the servers and whether
their configuration was loaded into Manager.
Description This column describes the type of server being detailed by the row. It also includes a
status indicator for the configuration file that Manager has loaded for the server.
• Green - Configuration Loaded The configuration of the server has been
successfully retrieved and can be edited in Manager.
• Yellow - Offline Configuration Loaded The configuration loaded is an offline
configuration. This will appear for a server that has been added to the solution
when the physical server is not currently connected on the network and Create
Offline Configuration was selected. The offline configuration file is stored on and
retrieved from the primary server until it can be replaced by or replace the actual
server configuration.
• Red - Configuration Not Loaded There is no configuration for the system
loaded even though the solution configuration includes an entry for the server.
This will appear for a server that has been added to the network when the
physical server is not currently connected on the network and Create Offline
Configuration was not selected. If may also appear if the server is currently not
contactable.
• Grey - No Connection This icon is used in conjunctions with the others to
indicate that there is no current connection to the server. For example:
- In conjunction with a green icon, it indicates that the server for which a
configuration has been loaded cannot be detected on the network. This may
be a temporary issue caused by that particular server rebooting following a
configuration change.
- In conjunction with a red icon, it indicates that the server for which a
configuration has not been loaded has now been detected on the network.
Saving and reloading the solution configuration may resolve the issue.
Name This is the server name as taken from its configuration file. Offline is shown if no
configuration file is available.
Table continues…

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 Server Edition Solution View

Address The IP address of the server. This is the address that is used when
Manager attempts to retrieve the servers configuration when loading the solution
configuration.
Primary Link This value indicates the configuration settings of the H.323 IP trunk between the
primary server and the server indicated by the row. It should state Bothway. If
it states anything other, that indicates a mismatch in H.323 IP trunk configuration
between the system and the primary server. To correct this, right-click on the row
and select Connect to Primary.
Secondary Link This column is only shown after a secondary server has been added to the
configuration of the solution. The value indicates the configuration settings of the
H.323 IP trunk between the secondary server and the server indicated by the row. It
should state Bothway. If it states anything other, that indicates a mismatch in H.323
IP trunk configuration between the system and the secondary server. To correct this,
right-click on the row and select Connect to Secondary.
Users Configured This column summarizes the number of users (other than NoUser) configured on
the server. A total for the whole network is shown in the Solution row.
Extensions Configured This column summarizes the number of extensions configured on the server. A total
for the whole network is shown in the Solution row.

Right-clicking on a server in the table may present a number of actions. The actions available vary
with the current state of the network configuration.
Option Description
Remove Remove the server from the solution configuration.
Connect to Primary Repair the configuration of the IP Office lines between the server and the primary server.
Connect to Repair the configuration of the IP Office lines between the server and the secondary
Secondary server.
Create Offline Create an offline configuration file for a server for which no actual configuration has
Configuration been loaded. The Offline Configuration menu is displayed followed by the Initial
Configuration menu for the server type. The offline configuration file is saved on the
primary server.

Related links
Server Edition Solution View on page 65

Opening additional applications

You can use the Open... option shown in the Solution View to view specific menus and to start a
number of additional applications.
Open... Description
Configuration Switch from the solution view to the configuration view.
System Status Launch the System Status Application application if installed on the same PC as IP
Office Manager.
Table continues…

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Working with the Server Edition Manager User Interface

Open... Description
Voicemail Launch the Voicemail Pro client application if installed on the same PC as IP Office
Administration Manager.
Resiliency Display the resiliency administration menu.
Administration
On-boarding Display the on-boarding menu used for new IP Office systems.
IP Office Web Launch IP Office Web Manager.
Manager
Help Access help for the solution view.

Related links
Server Edition Solution View on page 65

Setting all nodes to Subscription mode

If the primary server is configured as a subscription mode system, all other servers in the network
must also use subscription mode. To do that to convert the servers in an existing non-subscription
network:
1. Convert the primary server to subscription mode by rerunning the initial configuration menu
(File > Advanced > Initial Configuration).
2. In the solution view, use Set All Nodes to Subscription to change the mode of all other
servers in the network.
Related links
Server Edition Solution View on page 65

Setting all nodes to Select mode

Set All Nodes to Select
All systems in the solution must use the same licensing type. The Set All Nodes to Select
command shown in the Solution View is used implement Select licensing on all servers in the
network.
Related links
Server Edition Solution View on page 65

Setting the License Source in Server Edition View

All systems in the Server Edition solution must use the same license source. The license source is
defined by the configuration setting License | License | License Source.
The Set All Nodes License Source shown in the Solution View can be used to set all nodes to
use the same license source.
Related links
Server Edition Solution View on page 65

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 System Inventories

Adding a Server in the Server Edition Solution View

The Add option on the Solution View allows you to add another server. For example, add a Server
Edition Secondary Server or an Server Edition Expansion System.
When you add a system, IP Office Lines connecting the new system are configured with default
settings.
Important:
If the Manager setting File > Preferences > Preferences > SE Central Access is set to On,
an IP Office Line is not configured from the new system to the Server Edition Primary Server.
The status of the new system is Offline. You must configure an IP Office Line from the new
system to the Server Edition Primary Server.
Related links
Server Edition Solution View on page 65

System Inventories
Manager can be used to display a system inventory for any of the servers in the Server Edition
solution. The system inventory is a quick summary of key settings and information about the
server. It can also display an overview system inventory for the whole Server Edition solution.
Displaying a Server's System Inventory
The method for displaying the system inventory depends on what is currently being displayed by
Manager.
In the Server Edition Solution View, using the table at the bottom of the menu, click on the server
for which you want to display the system inventory. Click on Network for the inventory of the
Server Edition network.
or
In the navigation pane, click on the icon of the server for which you want to display the system
inventory. Click on the Network icon for the inventory of the Server Edition network.
Related links
Working with the Server Edition Manager User Interface on page 65

Default Settings
Most of the defaults for systems in a Server Edition solution match those of individual IP Office
systems as detailed in the Configuration Settings section. The table lists some differences.
All auto-create extension and auto-create user settings for IP devices are set to off.

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Working with the Server Edition Manager User Interface

Settings Primary Server Secondary Server Expansion System

System Time Settings Hidden. Time taken from SNTP from the primary server.
host server.
Voicemail Voicemail Pro Centralized Voicemail to the primary server
Alarms Syslog relay all alarms Syslog relay all alarms to the primary server.
to the local host.
IP Address Specified during initial configuration menu.
Lines Physical – – Auto-created
IP Lines H.323 line to the H.323 line to the H.323 line to the
secondary and each primary and each primary and to the
expansion system. expansion system. secondary if present.
Backup to secondary. Backup to primary. No backup.
Extension Physical – – Auto-created but no
base extension setting.
IP None None None
User None None None
Hunt Group None Not allowed Not allowed
Incoming Call Route None Replicated from primary.
Directory Stored on the primary Configured to obtain system directory from the
primary server.
User Rights None Replicated from primary.
ARS 50:Main Short code to secondary Short code to primary Short code to primary
and to secondary if
present.

Related links
Working with the Server Edition Manager User Interface on page 65

Record Consolidation
By default, to maintain the configurations of the systems in a Server Edition solution in synch,
certain types of configuration records are consolidated. That is, they are replicated in the individual
configuration of each system in the network. Consolidation is applied to:
• Short Codes - System short codes only.
• Time Profiles
• Account Codes
• User Rights
• Locations - Even when consolidated, the Emergency ARS and Fallback System settings
for each location are configured individually on each system.
Consolidate Network Operation

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 Telephone Features Supported Across Server Edition and SCN Networks

Use of consolidate settings is controlled by the File > Preferences > Preferences > Consolidate
Solution to Primary Settings setting.
Setting Description
Enabled • Entry and administration of consolidated records is performed only at the solution level.
• Those records are then automatically replicated in the configurations of all the systems
in the solution but, except for locations, are still only visible and editable at the solution
level.
• When the configurations are loaded or when this setting is changed to become
selected, if any inconsistency between records are found, a Consolidation Report
is displayed. This report allows selection of whether to update the system to match the
primary or to update the primary to match the system.
Disabled • Entry and administration of consolidated records can be performed at both the solution
and individual system levels.
• Records entered and edited at the solution level are still automatically replicated in the
configurations of all the systems in the solution. Each record displays a label on the
record indicating that it is a record that is shared across the solution.
• If a shared record is edited at the individual system level, that copy of the record is
no longer shared with the other systems. It will not be updated by any changes to the
solution level version of the same record.
• No consolidation checking for inconsistencies is done when the configurations are
loaded.

Related links
Working with the Server Edition Manager User Interface on page 65

Telephone Features Supported Across Server Edition and

SCN Networks
In addition to sharing user and hunt group extension numbers, the following additional telephony
features are supported between systems in a multi-site network. Features not listed are not
supported across the multi-site network.
• Absence Text
• Advertised Hunt Groups Hunt groups set to advertised can be dialed by users on other
systems
• Anti-tromboning Calls routed across the multi-site network and back to the originating
system are turned back into internal calls on the originating system only.
• Break Out Dialing
• Call Park / Unpark Call
• Call Pick-up Extension

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Working with the Server Edition Manager User Interface

• Call Tagging
• Callback When Free
• Centralized Call Log
• Centralized Personal Directory
• Conference
• Distributed Hunt Groups
• Distributed Voicemail Server Support
When using Vociemail Pro, each system can support its own Voicemail Pro server.
• Enable ARS / Disable ARS
• Extension Dialing
Each system automatically learns the user extension numbers available on other systems
and routes calls to those numbers.
• Resiliency Options
• Fax Relay
• Follow Me Here / Follow Me To
• Forwarding
• Hold
Held calls are signalled across the network.
• Internal Twining
• Intrusion Features
• Mobile Call Control
Licensed mobile call control users who remote hot desk to another system take their licensed
status with them.
• Music On Hold Source Selection
• Remote Hot Desking
• Set Hunt Group Out of Service / Clear Hunt Group Out of Service
• Transfer
Calls can be transferred to network extensions.
• User DSS/BLF
Monitoring of user status only. The ability to use additional features such as call pickup
via a USER button will differ depending on whether the monitored user is local or remote.
Indication of new voicemail messages provided by SoftConsole user speed dial icon is not
supported.
• User Profile Resilience

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 Telephone Features Supported Across Server Edition and SCN Networks

When a user hot desks to another system, they retain their Profile settings and rights.
Related links
Working with the Server Edition Manager User Interface on page 65
Small Community Networking on page 809

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Part 2: Menu Bar Commands

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Chapter 6: Menu Bar Commands

The commands available through the Manager's menu bar change according to the mode in which
Manager is running. Commands may also be grayed out if not currently applicable. For some
commands, an arrow symbol indicates that there are sub-commands from which a selection can be
made.
The following sections outline the functions of each command. The Edit and Help menus are not
included.

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Chapter 7: File Menu

Related links
File > Open Configuration on page 76
File > Close Configuration on page 77
File > Save Configuration on page 77
File > Save Configuration As on page 78
File > Change Working Directory on page 79
File > Preferences on page 81
File > Offline on page 81
File > Advanced on page 82
File > Backup/Restore on page 82
File > Import/Export on page 82
File > Exit on page 82

File > Open Configuration

This command displays the Select IP Office window used to receive a systems configuration
settings.

The same action is performed by the icon in the Main Toolbar.

The Select IP Office menu is also used for other actions such as reboot and sending a
configuration. If the unit required is not found, the Unit/Broadcast Address can be changed
and then Refresh clicked. To change the TCP addresses scanned, select File | Preferences |
Discovery and enter the required addresses in the IP Search Criteria.
Known Units is not available unless configured
Related links
File Menu on page 76

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 File > Close Configuration

File > Close Configuration

This command closes the currently loaded configuration without saving it.
Related links
File Menu on page 76

File > Save Configuration

The File > Save Configuration command saves the amended configuration.
• If the configuration has been received from a system, the Send Config menu is displayed.
• If the configuration file has been opened offline or created from new, the file is saved to disk
only.
Related links
File Menu on page 76
Saving Server Edition Configuration Changes on page 77

Saving Server Edition Configuration Changes

After making configuration changes using Manager, the process of saving the changes performs
several actions:
• The configurations are validated for consistency.
• For those systems for which configuration changes have been made, the new configuration is
sent to that system.
Manager defaults to rebooting those systems where a reboot is necessary for the configuration
changes to come into effect.
A time stamped copy of the new configuration is also stored on the Primary Server server.
• For a new Secondary Server or expansion system added to the network configuration using
the create off-line configuration option, the offline file is stored, allowing the new system to be
configured even though not yet physically present.
• When opening the configuration from a network, if the timestamp of the stored copy differs
from that of the actual system configuration, Manager will prompt for which configuration it
should load for editing.
Saving Configuration Changes
1. Click in the main toolbar or select File | Save Configuration from the menu bar.
2. The menu displayed only shows details for those systems where the system configuration
has been changed and needs to be sent back to the system.

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File Menu

Option Description
Select By default all systems with configuration changes are selected. If you want to exclude
a system from having its configuration updated, either deselect it or cancel the whole
process.
Change Mode If Manager thinks the changes made to the configuration settings are mergeable, it will
select Merge by default, otherwise it will select Immediate.
Merge Send the configuration settings without rebooting the system. This mode should only be
used with settings that are mergeable.
Immediate Send the configuration and then reboot the system.
When Free Send the configuration and reboot the system when there are no calls in progress. This
mode can be combined with the Incoming Call Barring and Outgoing Call Barring
options.
Store Offline It is possible to add a reference for a Server Edition Secondary or for a Server Edition
Expansion System to create a configuration file for that system even though it is not
physically present. Store Offline saves that configuration on the Server Edition Primary in
its file store. The same file is retrieved from there until the physical server is present at
which time you are prompted whether to use the stored file or the actual servers current
configuration.
Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Incoming Call Barring and Outgoing Call Barring options.
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for the
system reboot. If the time is after midnight, the system's normal daily backup is canceled.
Incoming Call This setting can be used when the reboot mode When Free or Timed is selected. It bars
Barring the receiving of any new calls.
Outgoing Call This setting can be used when the reboot mode When Free or Timed is selected. It bars
Barring the making of any new calls.

Click OK. The progress of the sending of each configuration is displayed.

Related links
File > Save Configuration on page 77

File > Save Configuration As

The File > Save Configuration As command allows you to save a configuration a file on the
Manager computer. Note that dynamic configuration data, for example hunt groups advertised
from other systems in a network, are not included in a configuration file saved onto PC and then
reopened.
The command displays the Save As window. Select the drive and directory. and then enter the
new file name. Once you have entered the file name, the Save Configuration File window opens,
where you have the option to encrypt the file by entering a password. Leave the password blank if
you do not want to encrypt the file.

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 File > Change Working Directory

Important:
Encrypted configuration files can only be opened with Manager 9.1 or later. In earlier versions
of Manager, the file will open but it is empty.
Configurations saved onto the PC in this way can be reopened using the icon or the File >
Offline > Open File command. If the file has been encrypted, you must enter the password.
When Manager is running in Server Edition mode, the Save command operates differently.
Multiple files are saved, one .cfg file for each server in the network plus a single .cfi file
for the whole network.
The .cfi file can be used with the File > Offline > Open File Set command to open the whole set
of files in a single action.
Related links
File Menu on page 76

File > Change Working Directory

These fields set the default location where Manager will look for and save files.

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File Menu

Directory Description
Working Directory (.cfg This folder path is used for the following:
files)
• For SD card management functions, it sets the path for the MemoryCards
sub-folder used for actions such as SD card recreation.
• If either of the Save Configuration File After Load or Backup Files on Send
settings (see Security on page 90) are enabled, it sets the directory into
which Manager saves the .cfg and .bak files if enabled.
The default folder used depends on:
• the version of IP Office Manager
• whether it was installed from the full admin suite or from admin lite.
• whether it was installed and run with Windows administrator rights or not.
If installed and run with Windows administrator rights:
File Path
Pre-R11.1 FP2 C:\Program Files (x86)\Avaya\IP
SP3 Office\Manager
Full Admin Suite C:\Program Files (x86)\Avaya\IP Office
Admin Suite\Manager
Admin Lite C:\Program Files (x86)\Avaya\IP Office
Lite\Manager

If installed and run without Windows administrator rights: To show files

in the following folders, Show hidden files must be enabled in Windows file
explorer.

File Path
Pre-R11.1 FP2 C:\Users\<user_name>\AppData\Local\VirtualS
SP3 tore\Program Files (x86)\Avaya\IP
Office\Manager
Full Admin Suite C:\Users\<user_name>\AppData\Local\VirtualS
tore\Program Files (x86)\Avaya\IP Office
Admin Suite\Manager
Admin Lite C:\Users\<user_name>\AppData\Local\VirtualS
tore\Program Files (x86)\Avaya\IP Office
Lite\Manager

Table continues…

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 File > Preferences

Directory Description
Binary Directory (.bin files) Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.

Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.

Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records
to be removed.

Related links
File Menu on page 76

File > Preferences

See File Preferences on page 84.
Related links
File Menu on page 76

File > Offline

See File Offline on page 94.
Related links
File Menu on page 76

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File Menu

File > Advanced

See File Advanced on page 96.
Related links
File Menu on page 76

File > Backup/Restore

Backup Binaries and Configurations
This command copies all configuration files (.cfg) and software binary files (.bin) stored in
Manager's working directory to a selected folder.
Restore Binaries and Configurations
This command copies all configuration files (.cfg) and software files (.bin) stored in a selected
folder to the Manager's working directory.
Related links
File Menu on page 76

File > Import/Export

For full details of importing and exporting settings, see Importing and Exporting Settings on
page 190.
Action Description
Export This command allows you to export the selected parts of the configuration as either a set
of CSV text files (.csv) or a single binary file (.exp).
Import This command allows you to import configuration settings. Two formats are supported.
Binary files (.exp) are settings previously exported from a system. CSV text files (.csv)
can also be exported from a system or can be created using a plain text editor.

Related links
File Menu on page 76

File > Exit

The File | Exit command exits the Manager application.

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 File > Exit

Related links
File Menu on page 76

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Chapter 8: File > Preferences

This command displays a window for configuring various aspects of how the IP Office Manager
application operates. The window is divided into a number of tabs.
Related links
Preferences on page 84
Directories on page 86
Discovery on page 88
Visual Preferences on page 89
Security on page 90
Validation on page 92

Preferences
This tab is accessed through File | Preferences and then selecting the Preferences tab.
Setting Description
Edit Services Base TCP Default = Off
Port:
This field shows or hides the base communication port settings.
Service Base TCP Port Default = 50804.
Access to the configuration and security settings on a system requires Manager
to send its requests to specific ports. This setting allows the TCP Base Port used
by Manager to be set to match the TCP Base Port setting of the system. The
system's TCP Base Port is set through its security settings.
Service Base HTTP Port Default = 80.
Access to the HTTP server on a system requires Manager to send its requests
to specific ports. This setting allows the HTTP Base Port used by Manager to
be set to match the HTTP Base Port setting of the system. The system’s HTTP
Base Port is set through its security settings.
Enable Time Server Default = On.
This setting allows Manager to respond to RFC868 Time requests from systems.
It will provide the system with both the UTC time value and the local time value
of the PC on which it is running.
Table continues…

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 Preferences

Setting Description
Enable BootP and TFTP Default = Off.
Servers
This setting allows Manager to respond to BOOTP request from systems for
which it also has a matching BOOTP record. It also allows Manager to respond
to TFTP requests for files.
Auto Connect on start up Default = On
If on, when Manager is started it will automatically launch the Select IP Office
menu and display any discovered systems. If only one system is discovered,
Manager will automatically display the login request for that system or load its
configuration if the security settings are default.
Set Simplified View as Default = Off
default
If on, the Manager will start in simplified view mode if no configuration is loaded.
Default to Standard Mode Default = Off
If on, when a configuration from a new or defaulted system running in
Basic mode is loaded, Manager will automatically convert the configuration to
Standard mode. Sending the configuration back to the system will restart it in
Standard mode. Only select this option if the only systems you expect to install
are Standard systems.
This setting does not affect existing systems with non-default configurations.
Use Remote Access Default = Off.
If selected, access to all the configurations of a multi-site network is allowed via
remote access to the primary server on the multi-site network. When selected,
an additional Use Remote Access check box option is displayed on the Select
IP Office menu when the Open with Server Edition Manager check box option
is selected or if Manager is already running in Server Edition mode.

Note:
To enable remote access, you must first configure an SSL VPN service
between each Server Edition system and the Avaya VPN Gateway (AVG).
For information, refer to the Deploying Avaya IP Office™ Platform SSL VPN
Services manual.
Consolidate Solution to This setting is used when managing a network based around Linux-based
Primary Settings primary and secondary servers such as Server Edition. When enabled, certain
records, such as system short code, are automatically matched across all IP
Officesystems in the network. See Record Consolidation on page 70.
Table continues…

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File > Preferences

Setting Description
SE Central Access Default = Off. Applies to Server Edition systems only.
If On, all Server Edition systems in the network obtain their configuration data
from a central location on the Primary Server. As a result, the display of
configuration changes is delayed until a synchronization process runs.
The synchronization process runs every 40 seconds. If the configuration change
requires a system restart, a refreshed configuration display is delayed until 40
seconds after system restart.
This setting can be used to drive configuration changes into expansion systems
when the expansion systems are not reachable through Manager and the only
accessible system is the Primary Server.
When enabled:
• When adding a new system to the solution, an IP Office Line is not configured
from the new system to the Server Edition Primary Server. The status of the
new system is Offline. You must configure an IP Office Line from the new
system to the Server Edition Primary Server.
• You cannot open configurations with a release number of 9.0.x or earlier.
• The following File > Advanced menu options are not available:
- System Shutdown
- Memory Card Command
• While no configuration is open, the following File > Advanced menu options
are greyed out:
- Erase Configuration (Default)
- Reboot
- Erase Security Settings (Default)
SE Central Access Port Default = 7070.
When SE Central Access is set to On, the port used for routing HTTPS
requests for configuration synchronization.

Related links
File > Preferences on page 84

Directories
These fields set the default location where Manager will look for and save files.

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 Directories

Directory Description
Working Directory (.cfg This folder path is used for the following:
files)
• For SD card management functions, it sets the path for the MemoryCards
sub-folder used for actions such as SD card recreation.
• If either of the Save Configuration File After Load or Backup Files on Send
settings (see Security on page 90) are enabled, it sets the directory into
which Manager saves the .cfg and .bak files if enabled.
The default folder used depends on:
• the version of IP Office Manager
• whether it was installed from the full admin suite or from admin lite.
• whether it was installed and run with Windows administrator rights or not.
If installed and run with Windows administrator rights:
File Path
Pre-R11.1 FP2 C:\Program Files (x86)\Avaya\IP
SP3 Office\Manager
Full Admin Suite C:\Program Files (x86)\Avaya\IP Office
Admin Suite\Manager
Admin Lite C:\Program Files (x86)\Avaya\IP Office
Lite\Manager

If installed and run without Windows administrator rights: To show files

in the following folders, Show hidden files must be enabled in Windows file
explorer.

File Path
Pre-R11.1 FP2 C:\Users\<user_name>\AppData\Local\VirtualS
SP3 tore\Program Files (x86)\Avaya\IP
Office\Manager
Full Admin Suite C:\Users\<user_name>\AppData\Local\VirtualS
tore\Program Files (x86)\Avaya\IP Office
Admin Suite\Manager
Admin Lite C:\Users\<user_name>\AppData\Local\VirtualS
tore\Program Files (x86)\Avaya\IP Office
Lite\Manager

Table continues…

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File > Preferences

Directory Description
Binary Directory (.bin files) Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.

Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.

Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records
to be removed.

Related links
File > Preferences on page 84

Discovery
These settings affect the Select IP Office menu used by Manager to discovery systems.

Setting Description
TCP and HTTP Default = On.
Discovery
This setting controls whether Manager uses TCP to discover systems. The addresses
used for TCP discovery are set through the IP Search Criteria field below.
NIC IP/NIC Subnet This area is for information only. It shows the IP address settings of the LAN network
interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to
add the address range it is part of to the IP Search Criteria. Note that if the address
of any of the Manager PC's NIC cards is changed, the Manager application should be
closed and restarted.
IP Search Criteria This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes,
for example 135.64.180.170 - 135.64.180.175.
Table continues…

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 Visual Preferences

Setting Description
UDP Discovery Default = On
This settings controls whether Manager uses UDP to discover systems.
Enter Broadcast IP Default = 255.255.255.255
Address
The broadcast IP address range that Manager should used during UDP discovery.
Since UDP broadcast is not routable, it will not locate systems that are on different
subnets from the Manager PC unless a specific address is entered.
Use DNS Selecting this option allows Manager to use DNS name (or IP address) lookup to
locate a system. Note that this overrides the use of the TCP Discovery and UDP
Discovery options above. This option requires the system IP address to be assigned
as a name on the users DNS server. When selected, the Unit/Discovery Address
field on the Select IP Office window is replaced by a Enter Unit DNS Name or IP
Address field.
SCN Discovery If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as
a single configuration. At least one of the systems in the Small Community Network
must be running Release 6.0 or higher software. See Configuring Small Community
Networking on page 809. This does not override the need for each system in the
Small Community Network to also be reachable by the TCP Discovery and or UDP
Discovery settings above and accessible by the router settings at the Manager
location.

Related links
File > Preferences on page 84

Visual Preferences
Setting Description
Icon size Sets the size for the icons in the navigation pane between Small, Medium or Large.
Multiline Tabs Default = Off.
In the details pane, for record types with more than two tabs, Manager can either use
buttons to scroll the tabs horizontally or arrange the tabs into multiple rows. This
setting allows selection of which method Manager uses.

Related links
File > Preferences on page 84

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File > Preferences

Security
Additional configuration information
For additional configuration information, see Security Administration on page 162. Also see Avaya
IP Office™ Platform Security Guidelines.
Configuration settings
Controls the various security settings of Manager. To control the security settings of the system,
see the information on Security mode.
All settings, except Secure Communications, can only be changed when a configuration has
been opened using a user name and password with Administrator rights or security administration
rights.
Setting Description
Request Login on Default = On
Save
By default a valid user name and password is required to receive a configuration from
a system and also to send that same configuration back to the system. Deselecting
this setting allows Manager to send the configuration back without having to renter
user name and password details. This does not apply to a configuration that has
been saved on PC and then reopened. This setting can only be changed when a
configuration has been opened using a user name and password with Administrator
rights or security administration rights.
Close Configuration/ Default = On.
Security Settings
When selected, the open configuration file or security settings are closed after being
After Send
sent back to the system. This is the normal default. This setting does not affect multi-
site network modes of Manager which always close the configuration after saving.
Before disabling this setting, you should recall that the configuration held by a running
system can be changed by actions other than Manager. For example, changes made
by users through phones. Keeping a configuration open in Manager for longer than
necessary increases the chances that the copy of the configuration differs from the
current configuration of the running system and will overwrite those changes when
sent back to the system.
Save Configuration Default = Off.
File After Load
When selected, a copy of the configuration is saved to Manager's working directory
(see Directories on page 86). The file is named using the system name and the
suffix .cfg. This local file can only be changed when a configuration has been
opened using a user name and password with Administrator rights.
Table continues…

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 Security

Setting Description
Backup Files on Send Default = Off.
If selected, whenever a copy of a configuration is sent to a system, a backup copy is
saved in Manager's working directory. See the notes above.
The file is saved using the system name, date and a version number followed by
the Backup File Extension as set below. This setting can only be changed when a
configuration has been opened using a user name and password with Administrator
rights.
Backup File Extension Default = .BAK
Sets the file extension to use for backup copies of system configurations generated
by the Backup Files on Send option above.
Number of Backup Default = Unlimited.
Files to keep
This option allows the number of backup files kept for each system to be limited. If set
to a value other then Unlimited, when that limit would be exceeded, the file with the
oldest backup file is deleted.
Enable Application Default = On.
Idle Timer (mins)
When enabled, no keyboard or mouse activity for 10 minutes will cause the Manager
to grey out the application and re-request the current service user password. This
setting can only be changed when a configuration has been opened using a user
name and password with Administrator rights or security administration rights.
Secure Default = On
Communications
When selected, any service communication from Manager to the system uses the
TLS protocol. This will use the ports set for secure configuration and secure security
access. It also requires the configuration and or security service within the system's
security configuration settings to have been set to support secure access. Depending
on the level of that secure access selected, it may be necessary for the Manager
Certificate Checks below to be configured to match those expected by the system
for configuration and or security service.

• When Secure Communications is set to On, a padlock icon is displayed at all

times in the lower right Manager status field.
• For Server Edition systems, Manager will always attempt to use secure
communications regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds,
Manager will offer to fallback to using unsecured communications.
Table continues…

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File > Preferences

Setting Description
Manager Certificate When the Secure Communications option above is used, Manager will process and
Checks check the certificate received from the system. This setting can only be changed
when a configuration has been opened using a user name and password with
Administrator rights or security administration rights. The options are:
• Low: Any certificate sent by the system is accepted.
• Medium: Any certificate sent by the system is accepted if it has previously been
previously saved in the Windows' certificate store. If the certificate has not been
previously saved, the user has the option to review and either accept or reject the
certificate.
• High: Any certificate sent by the system is accepted if it has previously been
previously saved in the Windows' certificate store. Any other certificate cause a log
in failure.
Certificate Offered to Default = none Specifies the certificate used to identify Manager when the Secure
IP Office Communications option is used and the system requests a certificate. Use the Set
button to change the selected certificate. Any certificate selected must have an
associated private key held within the store:
• Select from Current User certificate store - Display certificates currently in the
currently logged-in user store.
• Select from Local Machine certificate store.
• Remove Selection – do not offer a Manager certificate.

Related links
File > Preferences on page 84

Validation
By default Manager validates the whole configuration when it is loaded and individual fields
whenever they are edited. This tab allows selection of when automatic validation should be
applied to configuration files loaded into Manager.
Setting Description
Validate configuration Automatically validate configuration files when they are opened in Manager.
on open
Validate configuration Validate the whole configuration when OK is clicked after editing a record. For large
on edit configurations, disabling this option removes the delay caused by validating the
configuration after every edit.
Prompt for If selected, when saving or sending a configuration, a prompt is displayed asking
configuration whether the configuration should be validated. If validation is selected and error
validation on save or are found, the send or save process is canceled. This option is disabled if Validate
send configuration on edit is selected.

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 Validation

Related links
File > Preferences on page 84

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Chapter 9: File > Offline

The File > Offline menu allows the creation and use of IP Office system configuration files other
than the live system configuration.
Related links
Create New Config on page 94
Open File on page 94
Open File Set on page 95
Send Config on page 95
Receive Config on page 95

Create New Config

Used to create an offline configuration.
This command starts a dialog that allows you to create a default offline configuration by specifying
the system locales, the type of control unit and expansion modules and the trunk cards fitted. The
same action is performed by the icon in the Main Toolbar.
Related links
File > Offline on page 94

Open File
This command allows a configuration file stored on PC to be opened in Manager.
Related links
File > Offline on page 94

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 Open File Set

Open File Set

This command is only available when manager is running in Server Edition mode. It can be used
to load a set of files previously saved offline using the File > Save Configuration As command.
When selected, browse to the location of the saved .cfi file and associated .cfg files and select
the .cfi file.
Related links
File > Offline on page 94

Send Config
This command is used to send an offline configuration to a system.

Warning:
• After this command is completed, the system is rebooted. This will end all calls and
services in progress.
After sending the configuration, you should receive the configuration back from the system and
note any new validation errors shown by Manager. For example, if using Embedded Voicemail,
some sets of prompt languages may need to be updated to match the new configurations locale
setting using the Add/Display VM Locales option.
Related links
File > Offline on page 94

Receive Config
This command displays the Select IP Office menu used to receive a systems configuration
settings.
Once the configuration has been received, you are prompted to save it on the PC.
Related links
File > Offline on page 94

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Chapter 10: File > Advanced

The File > Advanced menu provides access to the following commands.
Related links
Erase Configuration on page 96
Reboot on page 97
System Shutdown on page 97
Upgrade on page 98
Change Mode on page 101
Audit Trail on page 101
Security Settings on page 102
Erase Security Settings (Default) on page 102
Embedded File Management on page 103
Format IP Office SD Card on page 103
Recreate IP Office SD Card on page 105
Memory Card Command on page 106
Launch Voicemail Pro on page 107
System Status on page 107
LVM Greeting Utility on page 107
Generate WebLM ID on page 107
Initial Configuration on page 108
Add/Display VM Locals on page 112

Erase Configuration
This command returns the configuration settings of a system back to their default values. It does
not affect the system's security settings or audit trail record.
• This command is grayed out when SE Central Access is enabled.
When this command is used, the Select IP Office menu is displayed. Once a system is selected,
a valid configuration user name and password are required to complete the action.

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 Reboot

IP500 V2 systems using IP Office A-Law or IP Office U-Law System SD cards will default to
Basic Edition mode. Loading the configuration will switch Manager to simplified view. To change
the system back to operating in Standard mode, use Change Mode on page 101.
Related links
File > Advanced on page 96

Reboot
When this command is used, the Select IP Office window is displayed. Once a system is
selected, a valid user name and password are required. The type of reboot can then be selected
in the Reboot window.
• This command is grayed out when SE Central Access is enabled.
When the reboot occurs can be selected as follows:
Setting Description
Immediate Send the configuration and then reboot the system.
When Free Send the configuration and reboot the system when there are no calls in progress. This
mode can be combined with the Call Barring options.
Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for the
reboot. If the time is after midnight, the system's normal daily backup is canceled.
Call Barring These settings can be used when the reboot mode When Free is selected. They bar the
sending or receiving of any new calls.

Related links
File > Advanced on page 96

System Shutdown
This command can be used to shutdown IP500 V2 systems. The shut down can be either
indefinite or for a set period of time after which the system will reboot. For Linux based systems,
use the service commands in IP Office Web Manager
• This command is not shown when SE Central Access is enabled.
Warning:
• A shutdown must always be used to switch off the system. Simply removing the power
cord or switching off the power input may cause the loss of configuration data.

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File > Advanced

• This is not a polite shutdown, any user calls and services in operation will be stopped.
Once shutdown, the system cannot be used to make or receive any calls until restarted.
The shutdown process takes up to a minute to complete. When shutdown, the LEDs shown on the
system are as follows. Do not remove power from the system or remove any of the memory cards
until the system is in this state:
• LED1 on each IP500 base card installed will also flash red rapidly plus LED 9 if a trunk
daughter card is fitted to the base card.
• The CPU LED on the rear of the system will flash red rapidly.
• The System SD and Optional SD memory card LEDs on the rear of the system are
extinguished.
To restart a system when shutdown indefinitely, or to restart a system before the timed restart,
switch power to the system off and on again.
Once you have selected the system from the Select IP Office window, the System Shutdown
Mode window opens. Select the type of shutdown required:
• If a Timed shutdown is selected, the system will reboot after the set time has elapsed.
• If Indefinite is used, the system can only be restarted by having its power switched off and
then on again. For Linux based telephone systems, the telephony service must be restarted
through the server's web control pages.
Related links
File > Advanced on page 96

Upgrade
This command starts the Upgrade Wizard tool. The Upgrade Wizard is used to compare the
software level of the control unit and modules within systems against the software level of the .bin
binary files Manager has available. The Upgrade Wizard can then be used to select which units to
upgrade.
• Whilst the wizard shows Linux based systems, it is not used to upgrade them. Linux systems
are updated using IP Office Web Manager.

Warning:
• Incorrect use of the upgrade command can halt system operation and render units in the
system unusable. You must refer to the Technical Bulletins for a specific release for full
details of performing software upgrades to that release. There may be additional steps
required such as defaulting the security settings.
• Performing any other actions on a system during an upgrade or closing the upgrade
wizard and Manager during an upgrade may render systems unusable.
• During an upgrade the system may restrict calls and services. It will reboot and
disconnect all current calls and services.

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 Upgrade

• The Validate option must remain selected wherever possible. Use of unvalidated
upgrades is subject to a number of conditions outlined in the IP Office Installation Manual
and Technical Bulletins.
The list area shows details of systems found by the Upgrade Wizard and the software currently
held by those systems. The check boxes are used to select which units should be upgraded.
Upgrading will require entry of a valid name and password for the selected system.
Column Description
Name The name of the system as set in its configuration (System | System | Name) .
IP Address The IP address of the system.
Type The type of system and the names of the various firmware files used by external
expansion systems supported by the system type.
Version Details the current software each unit in the systems is running.
Edition Indicates the operation mode of the system.
Licensed Indicates the highest value software upgrade license present in the system's
configuration. The IP Office Release that is supported by that license is also indicated in
brackets.
Required License Indicates the software upgrade license required for the current level of software the
system is running. The IP Office Release that is supported by that license is also
indicated in brackets.
It does not refer to the software upgrade license required for the level of software which
is available for upgrade. The system must include a license for the specific level of
software it is required to run.
For IP500 V2 systems, a value of 255 indicates that the control unit is still in its initial 90
days where it can be upgraded to a higher level without requiring an upgrade license.
Available Shows the version of the matching firmware files that Manager has available (a –
indicates no file available) in its current working directory. Upgrading to a release higher
than that supported by the current Licensed level will leave the system unable to support
any functions until the appropriate upgrade license is added to the system configuration.

The Upgrade Wizard includes a number of check boxes that can be used to include other actions
as part of the upgrade process:
Option Description
Validate The Validate option should remain selected wherever possible. When selected, the
upgrade process is divided as follows: transfer new software, confirm transfer, delete
old software, restart with new software. If Validate is not selected, the old software is
deleted before the new software is transferred.
Backup System For any IP500 V2 systems being upgraded, the Backup system files option will cause
Files the system to backup its memory card files as part of the upgrade.
Table continues…

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File > Advanced

Option Description
Upload System File For any IP500 V2 system being upgraded, the Upload system files option will upload
various files:
• It copies the binary files for the system control unit and possible external expansion
modules.
• It copies the firmware files used by phones supported by the system.
• It copies the files for Web Manager.
• For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts
for those supported languages set as the system locale, user locales, incoming call
route locales and short code locales are upgraded.
Restart IP Phones This will cause those phone to load any upgrade phone firmware included in the system
upgrade (if using the system's memory card as their firmware file source).

Related links
File > Advanced on page 96

Searching for Systems

The default address used by the Upgrade Wizard is the address shown in the Manager title bar,
which is selected through File | Preferences. If the unit required is not found, the address used can
be changed.
Procedure
1. Enter or select the required address in the Unit/Broadcast Address field.
2. Click Refreshto perform a new search.

Changing the .bin File Directory

The directory in which the Upgrade Wizard looks for .bin files is set through Manager's Binary
Directory setting. This can be changed using Files | Change Working Directory or File |
Preferences | Directories.
Use this procedure to change it directly from the Upgrade Wizard.
Procedure
1. Right-click on the list area.
2. Select Select Directory.
3. Browse to and highlight the folder containing the .bin files. Click OK.
4. The list in the Available column will be updated to show the .bin files in the selected
directory that match units or modules listed.

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 Change Mode

Change Mode
This command can be used to change the operating mode of an IP500 V2 system from Basic
Edition to either standard IP Office or Server Edition expansion. To convert an existing system to
Basic Edition mode, use the default configuration options.
Important:
• Using this command will default the configuration. Therefore ensure that you have a
backup copy of the configuration before using this command in case it is necessary to
return to the previous mode.
• Do not use this command if the Default to Standard Mode option is enabled in the IP
Office Manager preferences. Disable the option first.
Note that if the system includes components not supported by the mode to which it is switched,
they will not work in the new mode. For example, ETR cards which are only supported in Basic
Edition.
In order to use this command, the system security settings must be at their default settings. The
current setting can be defaulted using the Erase Security Settings (Default) command.
After a mode change, the system restarts. If the system does not restart, the most likely cause is
that the systems security settings were not at their default settings.
Related links
File > Advanced on page 96

Audit Trail
The audit trail lists the last 16 actions performed on the system from which the configuration
loaded into Manager was received. It includes actions by service users such as sending a
configuration back, reboots, upgrades and defaulting the system.
Audit trail events can be output to a Syslog server through the system's System | System Events
settings.
The last failed action is always recorded and shown in red. It is kept even if there have been 16
subsequent successful actions.
The Audit Trail is part of the system configuration file received from the system. If the configuration
is kept open between send and reboot operations (ie. if Close Configuration/Security Setting After
Send is not selected), the Audit Trail will not show details of those operations. It will only show
details of those operations if the configuration is closed and then a new copy of the configuration
is received from the system.

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File > Advanced

Audit Details
When a specific access event is selected from the list, the following information is shown in the
Audit Details section:
• The Security User shows the service user name used for the access action.
• The Date and Time of Access indicate the local system time when the recorded event
occurred.
• The PC Login is the computer name of the PC used for the access.
• The PC IP Address and PC MAC Address are the IP address and MAC address of the PC
used for access.
• The Access Type details the type of action that was performed.
• The Outcome shows the system's response to the access. The outcome Success
(Warning) refers to the sending of a configuration that contains fields marked as errors
or warnings by Manager's validation function. Success (Clean) refers to the sending of a
configuration that does not contain any validation errors or warnings.
• The IP Office Firmware indicates the IP Office Release version.
Items Changed
The Items Changed area summarizes the changes contained in a sent configuration. Where
changes to a single record of a particular type are made, the Item Name field lists the individual
record changed. Where changes are made to several records of the same type, the Item Name
field displays Multiple items.
Related links
File > Advanced on page 96

Security Settings
This command is used to switch the Manager application to security mode. In that mode, Manager
is used to edit the security settings of a system.
Related links
File > Advanced on page 96

Erase Security Settings (Default)

This command returns the security settings of a system back to their default values. This action
does not affect the system's configuration or audit trail record. When this command is used, the
Select IP Office menu is displayed. Once a system is selected, a valid security user name and
password are required to complete the action.
• This command is grayed out when SE Central Access is enabled.

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 Embedded File Management

Note that any security certificates stored and being used by the system are deleted. Any services
currently using those certificates are disconnected and disabled until the appropriate certificates
are added back to the system's security configuration. That includes SSL VPN connections being
used to perform system maintenance.
The name and password used for this command are those required for security configuration
access which are different from those used for normal configuration access.
For IP500 V2 control units, if the security settings cannot be defaulted using this command, they
can be defaulted using a DTE cable connection to the system. Refer to the Deploying an IP500 V2
IP Office Subscription System manual for details.

Warning:
• Service Disruption - Whilst defaulting the security settings does not require a system
reboot, it may cause service disruption for several minutes while the system generates a
new default security certificate.
Related links
File > Advanced on page 96

Embedded File Management

For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files. See
Embedded File Management Menu on page 123.
• For non-Linux systems, the folders accessed using embedded file management are those on
the SD cards installed in the control unit.
• For Linux based systems, the folder path displays as disk in the embedded file manager.
That path maps onto /opt/ipoffice on the server.
Related links
File > Advanced on page 96

Format IP Office SD Card

This command allows suitable SD cards to be formatted by the Manager PC. The system supports
SD cards with the following format: SDHC minimum 4GB FAT32 format (Single partition, SDHC,
class2+, FAT32, SPI & SD bus). Non-Avaya supplied cards of the same format can be used a
system's Optional SD slot for additional actions such as backup.

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File > Advanced

Warning:
• Do not re-purpose a Enterprise Branch SD card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
• All File Will Be Erased
Note that this action will erase any existing files and folders on the card. If the
requirement is just to update the card, use Recreate IP Office SD Card without
reformatting. Once a card has been formatted, the folders and files required for operation
can be loaded onto the card from the Manager PC using the Recreate IP Office SD Card
command.
• Avaya supplied SD cards should not be formatted using any other method than the
format commands within Manager and System Status Application. Formatting the cards
using any other method will remove the feature key used for system licensing from the
card.
Related links
File > Advanced on page 96
Formatting the SD card on page 104

Formatting the SD card

Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File > Advanced > Format IP Office SD Card.
3. Select the type of card. This selection just sets the card label shown when viewing the card
details. It does not affect the actual formatting. Select the label that matches the file set you
will be placing on the card.
• IP Office A-Law - A system fitted with this type of card will default to A-Law telephony.
• IP Office U-Law - A system fitted with this type of card will default to U-Law telephony.
4. Browse to the card location and click OK.
5. The status bar at the bottom displays the progress of the formatting process.
6. When the formatting is complete, use the Recreate IP Office SD Card command to load
the system folders and files onto the card. See Recreate IP Office SD Card on page 105.
Related links
Format IP Office SD Card on page 103

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 Recreate IP Office SD Card

Recreate IP Office SD Card

This command can be used with the System SD cards used by IP500 V2 control units. It allows
Manager to copy all the files and folders used by a system when starting onto the card that has
been place into the card slot of the PC running Manager. It updates the card with the version
of those files installed with the Manager application. It includes the binary files for the system,
external expansion modules and phones. The command also copies all language prompt sets
used by Embedded Voicemail.
If the card contains dynamic system files such as SMDR records, they are temporarily backed up
by Manager and then restored after the card is recreated. For the card to be used in a system's
System SD slot the card must be Avaya SD Feature Key card. The card must be correctly
formatted, however a reformat of an existing working card is not necessary before using recreate
to update the card contents.
The source for the files copied to the SD card are the sub-folders of the
\Memory Cards folder under Manager's Working Directory (normally C:\Program
Files\Avaya\IPOffice\Manager). However, if the Working Directory is changed to a location
without an appropriate set of \Memory Cards sub-folders, the required set of files will not be
copied onto the SD card.
Related links
File > Advanced on page 96
Recreating the IP Office SD Card on page 105

Recreating the IP Office SD Card

About this task
Note:
This process can take up to 20 minutes depending on the PC. Once started, the process
should not be interrupted.
Memory cards folder is missing in IPOAdminLite.exe installation because of which you will
not be able to upgrade system files or Web Manager files using IPOAdminLite Manager. Also
you will not be able to recreate SD card with IPOAdminLite.exe installation.
Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File | Advanced | Recreate IP Office SD Card.
3. Select the type of system for which the card is intended.
This selection will affect how the system operates when defaulted with this card present in
its System SD card slot.
• IP Office A-Law A system fitted with this type of card will default to A-Law telephony.
• IP Office U-Law A system fitted with this type of card will default to U-Law telephony.

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File > Advanced

• Enterprise Branch Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch mode. There is a separate SD card for IP Office.
The Enterprise Branch SD card can only be used for IP Office operation and cannot be
used tochange modes to IP Office. You also cannot use or change an IP Office SD card
for use withan Enterprise Branch system.

Warning:
Do not re-purpose a Enterprise Branch SD card for use with any other IP Office
mode. Doing so may damage the SD card and make it unusable for your Enterprise
Branch system.
4. Browse to the card location and click OK.
5. For all systems, these files are necessary if you want to go through the process of on-
boarding registration.
6. Manager will start creating folders on the SD card and copying the required files into those
folders.
7. Do not remove the card until the process is completed and Manager displays a message
that the process has been completed.
Related links
Recreate IP Office SD Card on page 105

Memory Card Command

These commands are used with the memory cards installed in IP500 V2 control units.
• This command is not shown when SE Central Access is enabled.
Command Description
Shutdown This command can be used to shutdown the operation of IP500 V2 unit memory cards.
• This action or a system shutdown must be performed before a memory card is
removed from the unit. Removing a memory card while the system is running may
cause file corruption. Card services can be restarted by either reinserting the card or
using the Start Up command.
• Shutting down the memory card will disable all services provided by the card including
Embedded Voicemail if being used. Features licensed by the memory card will
continue to operate for up to 2 hours.
Start Up This command can be used to restart operation of an IP500 V2 memory card that has
been shut down. The command will start the Select IP Office discovery process for
selection of the system.

Related links
File > Advanced on page 96

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 Launch Voicemail Pro

Launch Voicemail Pro

If the Voicemail Pro client is installed on the same PC as Manager, this link can be used to launch
the Voicemail Pro client. This can also be done by clicking on the icon in the Manager toolbar.
Related links
File > Advanced on page 96

System Status
System Status is an application that can be used to monitor and report on the status of a system.
This is a separate application from Manager. If installed on the same PC, it can be started using
the File | Advanced | System Status link within Manager. Use of the application requires a
service user name and password configured on the system for System Status Access within the
system's security settings.
Related links
File > Advanced on page 96

LVM Greeting Utility

This command launches a utility that can be used to convert .wav files to the format used by
Embedded Voicemail (.c11). You can use the converted files for user mailbox name, greeting and
announcement prompts. You can also use them for auto-attendant prompts. For more information,
refer to the IP Office Embedded Voicemail Installation manual.
• The command is only selectable when the IP Office system's Voicemail Type is set as
Embedded Voicemail.
• The utility can be run separately using the file LVMGreeting.exe found in the
LVMGreeting sub-folder of the Manager application.
Related links
File > Advanced on page 96

Generate WebLM ID
This menu is only used for Linux based systems. Any system being upgrading from per-Release
10 ADI licenses must be migrated to PLDS licenses before upgrading. This is done using the files
created by the license migration tool. However, that tool assumes that the system will also be

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File > Advanced

the license host. If instead the system is going to use PLDS license hosted by a WebLM server,
the system's web license server host ID is required in addition to the files created by the license
migration tool. The Generate WebLM ID tool provides that additional ID.
To generate the server’s Web License Server Host ID:
1. Click File > Advanced > Generate WebLM ID. The menu displayed varies depending on
whether the server is virtualized or not.
2. Enter the details of the server. For a virtualized server, the UUID can be obtained from one
of the following:
• Using the command line command: dmidecode -s system-uuid
• From the uuid.bios line of the virtual machines vmx file.
• From the VSphere client. See http://www-01.ibm.com/support/docview.wss?
uid=swg21682150.
3. Click Generate.
Related links
File > Advanced on page 96

Initial Configuration
The Initial Configuration menu is displayed for all new or fully defaulted systems. It allows the
required operating mode for the system to be selected.
• For an existing system, you can re-run the initial configuration by selecting File >
Advanced > Initial Configuration.
• The Initial Configuration utility changes the security settings. Therefore, the user running
the utility must have security read/write rights.

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 Initial Configuration

Common Settings
Option Description
System Mode Sets the operating mode of the server. The options available depend on the type of server
platform. For further details, refer to the appropriate IP Office deployment manual.
• For Linux-based servers:
- Server Edition
- Server Edition - Select
- Server Edition - Subscription
• For an IP500 V2 server:
- IP Office Standard Edition
- IP Office Subscription
- IP Office ACO ATA Gateway
- Server Edition Expansion
- Server Edition Expansion - Subscription
• For an existing IP Office being reconfigured, the choice of system modes is restricted.
For example, you cannot change a subscription mode system to non-subscription
mode. In order display the full set of options, you must default the IP Office system
configuration .
System Name A name to identify this system. This is typically used to identify the configuration by
the location or customer's company name. Some features require the system to have a
name.
• This field is case sensitive and within any network of systems must be unique.
• Do not use <, >, |, \0, :, *, ?, . or /.
Retain This option is shown for existing servers where the initial configuration menu is being
Configuration rerun.
Data
• If cleared, the existing configuration of the IP Office system is defaulted.
• If enabled, the existing configuration is retained. However, some elements of that
configuration may be invalid or ignored. It is your responsibility to ensure that the final
configuration is valid.
Locale This setting sets default telephony and language settings based on the selection. It
also sets various external line settings and so must be set correctly to ensure correct
operation of the system. See Avaya IP Office Locale Settings. For individual users, the
system settings can be overridden through their own locale setting (User > User >
Locale).
Table continues…

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File > Advanced

Option Description
Default Extension Default = Existing default extension password
Password
The field provides you with option to view and edit the existing default extension
password. The default extension password is set up during IP Office installation either
by the administrator or is randomly generated by the system. The system generated
random password is of 10 digits. Use the Eye icon to see the existing default password.
The password must be between 9 to 13 digits.
Hosted This option is only used on non-subscription Server Edition system. If enabled, it indicates
Deployment that the system is a hosted deployment.
Services Device ID This setting is shown for Server Edition servers only. The ID is displayed on the Solution
view, System Inventory and on the System > System tab in the configuration.
• The value can be changed using the Device ID field on the System > System Events
configuration tab.

Subscription System Details

These details are only shown for subscription mode systems. They are used by the system to
obtain its subscriptions. They details required are supplied when the system if registered for
subscription.
Name Description
System ID This is a fixed value against which the system's subscriptions are issued and
validated.
• For an IP500 V2 system, this ID is based on the System SD card installed in the
system.
Customer ID The customer ID specified when the system was registered for subscriptions.
License Server Address The address of the server which provides the system with its subscriptions.

LAN Configuration Settings

Name Description
Public LAN Interface Select which of the server's LAN interfaces is connected to the customer network
routed to the external internet. Additional IP Route details are added to the system
configuration based on this choice.
Gateway The address of the default gateway on the customer network to which non-LAN
traffic should be routed.
After initial configuration, a default IP route is created, using this address and the
selected Public LAN Interface setting.
DNS Server The address used on the customer network for resolution of DNS queries. This is
either the customer's DNS server or the DNS address provided by their internet
service provider.
LAN1 CONFIGURATION/LAN2 CONFIGURATION
Separate sets of LAN configuration details are shown for LAN1 and LAN2.
Table continues…

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 Initial Configuration

Name Description
IP Address The base IP address for the LAN. The defaults are 192.168.42.1 for LAN1 and
192.168.43.1 for LAN2.
If the server is acting as the DHCP server for the LAN, this is the starting address
for the DHCP address range.
IP Mask Default = 255.255.255.0. This is the IP subnet mask used with the IP address.
DHCP Mode Select whether the server performs DHCP for the LAN.
• Server - When this option is selected, the system will act as a DHCP Server on
this LAN, allocating address to other devices on the network and to PPP Dial in
users.
- Devices on requesting an address are allocated addresses from the bottom of
the available address range upwards.
- Dial In users are allocated addresses from the top of the available range
downwards.
- If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users
are allocated their address from the LAN1 pool of addresses first.
• Disabled - When this option is selected, the system will not use DHCP to get or
issue IP addresses.
• Dial In - When this option is selected, the system will allocate DHCP addresses
to PPP Dial In users only. On systems using DHCP pools, only addresses from a
pool on the same subnet as the system's own LAN address will be used.
• Client - When this option is selected, the system request its IP Address and IP
Mask from another DHCP server on the LAN.
Enable NAT Default = Off.
Shown for IP500 V2 systems only. This setting controls whether NAT should be
used for IP traffic from LAN1 to LAN2.

Solution Settings
These settings are shown for Linux-based systems. The options vary depending on the server's
role in the network (primary, secondary or expansion).
Name Description
Server Edition Primary For secondary and expansion servers, specify the address of the primary server.
Server
Server Edition For primary and expansion servers, specify the address of the secondary server.
Secondary Server
WebSocket Password For each of the addresses set above, a bi-directional WebSocket connection is
created. A matching password must be set at each end of the line.
DNS Server This is the IP address of a DNS Server. If this field is left blank, the system uses
its own address as the DNS server for DHCP client and forwards DNS requests
to the service provider when Request DNS is selected in the service being used
(Service > IP).

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File > Advanced

Centralized Management
The following settings are used for IP Office systems being deployed as branch systems in a
network managed using System Manager. Refer to the Deploying Avaya IP Office™ Platform as
an Enterprise Branch with Avaya Aura® Session Manager manual.
Name Description
Under Centralized When selected, the additional fields below are shown.
Management
SMGR Address Enter the IP address of the System Manager server managing the branch network.
Redundant SMGR Enter the IP address of the secondary System Manager server managing the
Address network.
SMGR Community The shared community name for servers within the branch network.
SNMP Device ID The unique SNMP ID for the IP Office server within the network.
Trap Community The public name for sending SNMP trap alarms.
SCEP Domain The domain name for SCEP (Simple Certificate Enrollment Protocol) operation in
Certificate Name the branch network.
Certificate Enrollment The password for requesting certificates from the network's SCEP server.
(SCEP) Password

Related links
File > Advanced on page 96

Add/Display VM Locals
This option is only displayed when the configuration from an IP500 V2 systems with its Voicemail
Type set to Embedded Voicemail is received in Manager. It is not shown for off-line configuration
or configurations loaded from a PC file.
Selecting this option displays a list of the Embedded Voicemail prompt languages. Those
languages already present on the System SD card or not supported are grayed out. Additional
languages can be selected and then uploaded from Manager to the system.
When editing the system configuration in Manager, if the locale language selected for the system,
a user, a short code or an incoming call route is not already present on the System SD card,
Manager will display an error. Add/Display VM locales can then be used to upload the prompts
for the required language in order to correct the error.
You can reload languages that are already installed on the System SD card. For example, you
may want to reload the languages if new prompts have been added in a maintenance release.
To reload existing languages, upgrade the system (File | Advanced | Upgrade) with the Upload
System Files option checked. You can also choose Upload System Files from the Embedded File
Management utility (File | Advanced | Embedded File Management).
The Recreate IP Office SD Card command can be used to locally load all available languages
onto an SD card. See Recreate IP Office SD Card on page 105.

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 Add/Display VM Locals

Related links
File > Advanced on page 96

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Chapter 11: View Menu

View > Toolbars

Allows selection of which toolbars should be shown or hidden in configuration mode. A tick mark is
displayed next to the name of those toolbars that are currently shown.
View > Navigation Pane
Shows or hides the Navigation Pane. A tick mark appears next to the command when the pane is
shown.
View > Group Pane
Shows or hides the Group Pane. A tick mark appears next to the command when the pane is
shown.
View > Details Pane
Sets the location of the Details Pane when the Group Pane is also shown. The Details Pane can be
placed either below or to the right of the Group Pane.
View > Error Pane
Shows or hides the Error Pane. A tick mark appears next to the command when the pane is shown.
View > Advance View
Causes Manager to switch from its simplified view to advanced view mode. Manager automatically
switches to advanced view mode if a Standard Edition configuration is loaded.
View > Simplified View
If Manager has no configuration loaded, this command switches it from advanced view to simplified
view.
View > TFTP Log
This command displays the TFTP Log window. This window shows TFTP traffic between Manager
and devices that uses TFTP to send and receive files. For example, the TFTP Log below shows an
Avaya IP phone requesting and then being sent its software files.

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Chapter 12: Tools Menu

Related links
Tools > Extension Renumber on page 115
Tools > Line Renumber on page 115
Tools > Connect To on page 116
Tools > Export > User on page 116
Tools > SCN Service User Management on page 117
Tools > Busy on Held Validation on page 118
Tools > MSN Configuration on page 118
Tools > Print Button Labels on page 118
Tools > Import Templates on page 119
File > Advanced > Generate WebLM ID on page 120
Tools > License Migration on page 120

Tools > Extension Renumber

This command allows the extension numbering of user extensions to be changed. The existing
extension number range to be adjusted can be specified followed by the new start point for the
range after renumbering.
The command does not alter the extension number used for hunt groups but does adjust the
extension numbers of hunt group members.
Related links
Tools Menu on page 115

Tools > Line Renumber

On external trunks Line appearance ID numbers can be assigned to each channel supported in
order to allow that channel or line to be associated with a Line Appearance button on phones that
support button programming. By default all lines are automatically numbered from 701 upwards
when added to the system. This command allows the lines to be renumbered from a different
starting point.

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Tools Menu

Related links
Tools Menu on page 115

Tools > Connect To

This option can be used to create H.323 IP line connections between two systems in a multi-site
network, one being the system with its configuration currently loaded in Manager, the other being
selected from a discovery dialog. This option is not available for Server Edition mode.
Important:
This process will require the systems to be rebooted.
Procedure
1. With the configuration of the first system received from that system and displayed in
Manager, clicking on or Tools | Connect To
2. A discovery menu is displayed and will list any other systems discovered.
3. Select the system to which connection is required.
4. Enter the login name and password for configuration access to that system.
5. Manager will switch to Small Community Network management mode, displaying the
configuration of both systems.
6. Click to save the new configuration back to each system.
Related links
Tools Menu on page 115

Tools > Export > User

This tool is used for systems being migrated to a enterprise environment managed from Avaya
System Manager. Refer to the Migrating an IP Office or B5800 Branch Gateway to an IP Office
Enterprise Branch manual for full details of the process. If you want to export a list of users and
extension numbers for other purposes use File > Import/Export.
When performing such a migration, users are not automatically created in System Manager. The
existing users must be exported to an XML file and then that file must be imported into System
Manager.
This feature lets you to export all or selected users from the current configuration to an XML file.
The default filename is <SystemName>_Users.xml.
The Domain Name field is added to the user names as part of the user identity in the XML file.

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 Tools > SCN Service User Management

Related links
Tools Menu on page 115

Tools > SCN Service User Management

When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This tool is also available within a Server Edition network for administration of the Administrator
account.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Procedure
1. The Select IP Office menu displays the list of discovered systems.
2. Select the systems for which you want to create a common configuration account.
Click OK.
3. A user name and password for security configuration access to each system is requested.
Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials
for all remaining, selected IPOs.
4. The systems will be listed and whether they already have an SCN_Admin account is
shown.
5. To create the SCN_Admin account on each system and set the password for those
account click on Create Service User.
6. Enter the common password and click OK.
7. The password can be changed in future using the Change Password option.
8. Click Close.
Related links
Tools Menu on page 115

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Tools Menu

Tools > Busy on Held Validation

Busy on Held is a user feature where, when the user has a call on hold, the system indicate the
user as being busy to any further calls.
The use of Busy on Held in conjunction with multiple call appearance buttons is deprecated. This
command can be used to identify those users who have multiple call appearance buttons and for
whom Busy on Held is currently set.
When run, it shows a list of the users affected and if selected their Busy on Held setting will be
switched off.
Related links
Tools Menu on page 115

Tools > MSN Configuration

Used to populate the Incoming Call Route table with a range of MSN or DID numbers.
Setting Description
MSN/DID The first number in the set of MSN numbers for which you have subscribed.

Note:
If you require to find an exact match between the MSN numbers and the
destination numbers, enter a minus (-) sign before the first MSN number.
Destination Where incoming calls with matching digits should be routed. The drop-down list
contains the extensions and groups on the system.
Line Group ID Specifies the incoming line group ID of the trunks to which the DID routing is applied.
Presentation Digits Set to match the number of digits from the MSN/DID number that the central office
exchange will actually present to the system.
Range How many MSN or DID number routes to create in sequence using the selected
MSN/DID and Destination as start points. Only routing to user extensions is supported
when creating a range of records.

Related links
Tools Menu on page 115

Tools > Print Button Labels

This option is only enabled if a version of DESI software is also installed on the same PC as
Manager. It can then be used when a system configuration is loaded in Manager.

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 Tools > Import Templates

DESI software can be obtained from the Avaya support web site (http://support.avaya.com) or
from DESI (http://www.desi.com). Currently, though all users are shown, only ETR, M Series,
T-Series, 1400 and 1600 phones are supported by DESI templates.
The text used on the labels:
• If a text label has been added in the user's Button Programming settings, that text label is
passed to the DESI application.
• Note that the DESI application cannot import non-ASCII characters and may render them
incorrectly.
• Manager will display a warning if it estimates that the user's current text for some buttons
may exceed the label space of the phone type.
• If no text label has been set, the default label for the action currently assigned to the button is
passed to the DESI application.
• Once the labels are shown in the DESI application, the label text can be changed.
1. Load the configuration of the system for which you want to print button labels.
2. Select Tools and then Print Button Labels.
• Name/Extn – These are the user name and extension number details of the users in the
system configuration currently loaded in Manager.
• Phone Type – This field shows the type of phone, if known, that the user is currently
associated with. The drop down can be used to change the selection if required.
• Expansion Modules – If the phone type supports additional button modules, this drop
down can be used to select the type and number of button modules.
• Print Extn – This check box is used to select whether the phone button details should
be included in the output passed to the DESI software.
• Print BM1/Print BM2/Print BM3 – These check boxes are used to select whether
button module button details should be included in the output passed to the DESI
software. These button will only be selectable if the user's Expansion Modules is set to
the number of button modules.
3. Click Print via DESI to transfer the information to the DESI application. Within DESI, edit
the labels as required and then print the labels.
Related links
Tools Menu on page 115

Tools > Import Templates

Manager can be used to import and use templates. See Working with Templates on page 686.
Templates are stored in a default location. This command can be used to select a folder containing
other template files and copy those files into the correct default folder.

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Tools Menu

Related links
Tools Menu on page 115

File > Advanced > Generate WebLM ID

This menu is only used for Server Edition systems. Any system being upgrading from pre-Release
10 ADI licenses must be migrated to PLDS licenses before upgrading. This is done using the files
created by the license migration tool. However, that tool assumes that the system will also be the
licence host. If instead the system is going to use PLDS licenses hosted by a WebLM server,
the system's web license server host ID is required in addition to the files created by the license
migration tool. This tool generates that additional ID.
To generate the server’s Web License Server Host ID:
1. Click File > Advanced > Generate WebLM ID. The menu displayed varies depending you
indicate the server is virtualized or not.
2. Enter the details of the server:
• UUID: For a virtualized server, the UUID can be obtained as follows:
- Using the command line command: dmidecode -s system-uuid
- From the uuid.bios line of the virtual machines vmx file.
- From the VSphere client, see http://www-01.ibm.com/support/docview.wss?
uid=swg21682150.
3. Click Generate.
4. The system’s host ID is displayed. Copy and paste this value to a text file.
Related links
Tools Menu on page 115

Tools > License Migration

IP Office release 10 and higher supports only the Product Licensing and Delivery System (PLDS)
to manage license files. If you are upgrading from a previous release, you must migrate all of
your pre-R10 licenses (ADI, PLDS, mix of ADI/PLDS, virtual) to R10 PLDS licenses. The license
migration tool extracts all the licensing information from an IP Office system and saves it to a
file. This file can then be used prepare a software upgrade quote in the Avaya One Source
Configurator in order to obtain the required new PLDS R10 licenses.
For Server Edition deployments, the License Migration tool collects licensing information from
every node in the solution.
For a step by step procedure, see Migrating Licenses to PLDS on page 684.

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 Tools > License Migration

Notes
• Ensure all licenses are loaded on the system before using the license migration tool to
extract the licensing information. For Server Edition deployments, ensure all nodes are
online in order to capture the current view of systems in the solution. This step must be
completed before preparing your R10 software upgrade quote in the the Avaya One Source
Configurator.
• License migration is supported on all IP Office modes, release 6.0 and higher.
• The license migration tool can only be used with an online configuration. The Tools >
License Migration option is disabled for offline configurations.
• The generated file can be read but must not be edited. License migration will fail if the file has
been edited.
Note:
The License Migration Tool is not used for the upgrade of an SMGR WebLM license used
in certain pre-R10 Enterprise Branch deployments. For more information see the Avaya One
Source Configurator.
Related links
Tools Menu on page 115

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Chapter 13: Security Mode Menus

These commands are available when the Manager is in security configuration mode.
Security Settings > File > Open Security Settings
Displays the Select IP Office menu to select and load a system's security settings. This requires
entry of a user name and password with rights to access security settings of the selected system.
This behavior changes when configuration settings have already be received from a system using a
service user name and password that also has security access rights for that system. In that case,
the system's security settings are automatically loaded without requiring name and password entry.
Security Settings > File > Close Security Settings
Close the currently open set of security settings received from a system without saving those
settings.
Security Settings > File > Save Security Settings
Send edited security settings back to the system. Requires re-entry of a service user name and
password with access rights for security settings.
Security Settings > File > Reset Security Settings
Reset the security settings of the selected system to defaults. Requires entry of a service user name
and password with access rights for resetting the security settings. This option is not usable while a
set of security configuration settings is loaded.
The command File | Advanced | Erase Security Settings (Default) performs the same action from
Manager configuration mode.
Security Settings > File > Preferences
Displays a window for configuring various aspects of Manager's operation. The window is divided
into a number of tabs.
For a description of the Preferences options, see File > Preferences.
Security Settings > File > Configuration
Returns Manager to configuration mode.
Security Settings > File > Exit
This command closes Manager.

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Chapter 14: Embedded File Management
Menu

For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files.
• For non-Linux systems, the folders accessed using embedded file management are those on
the SD cards installed in the control unit.
• For Linux based systems, the folder path displays as disk in the embedded file manager. That
path maps onto /opt/ipoffice on the server.
• Access to embedded file management requires security right group permission for Rights
Groups > Configuration > Manager Operator Rights set to Administrator or Manager.
Embedded Voicemail Files
When viewing the memory card, the files related to Embedded Voicemail are visible, however these
files are greyed out (ie. cannot be deleted, downloaded or overwritten).
• Mailbox greetings and messages are shown as .clp files.
• The language prompts for Embedded Voicemail functions are stored in separate language
sub-folders of lvmail. These are .c11 files.
• Named prompt files for use by Embedded Voicemail auto attendants are stored in the
lvmail\AAG folder and use the same .c11 or .c23 file formats as the language prompts.
These files can be created from standard .wav files before being downloaded to the memory
card by using the LVM Greeting Utility.
Avaya IP Phone Files
The memory card can be used as the source of files requested by IP Phones when rebooting. For
phones using system DHCP, once the files are loaded onto the card, the TFTP Server IP Address
and HTTP Server IP Address on the System | System tab must be set to match the system's LAN
address.
Viewing a Memory Card
When Advanced | Embedded File Management is selected, the Manager will go through normal
system discovery but will only allow selection of systems which can support a memory card. When a
system is selected, a valid service user name and password for configuration access to that system
is requested. If the system selected does not have a memory card installed, the files view remains
blank and the message TFTP:Received TFTP Error "Not Found" appears in Manager's status bar.

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Embedded File Management Menu

Commands
Command Description
Open File Select a system and display the contents of its memory cards if any are present and in use.
Settings
Close File Close the current memory card contents listing without exiting embedded file management
Settings mode.
Refresh File This command can be used to request a file update from the system.
Settings
Upload File This command can be used to select and upload a file to the memory card in the system.
Upload System This command is available with IP500 V2 systems. When this command is selected,
Files Manager will upload the software files for operation to the System SD card.

Warning:
• After this command is completed, the system is rebooted. This will end all calls and
services in progress.
• It copies the binary files for the system control unit and possible external expansion
modules.
• It copies the firmware files used by phones supported by the system.
• For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts
for those supported languages set as the system locale, user locales, incoming call route
locales and short code locales are upgraded. In addition the English language prompts
are upgraded as follows: IP Office A-Law/Norstar SD Cards - UK English, IP Office U
Law/PARTNER SD Cards - US English.
Backup System This command is available with IP500 V2 systems. When selected, Manager copies the
Files folders and files from the System SD card's /primary folder to its /backup folder. Any
matching files and folders already present are overwritten. This action can be included as
part of the system's automatic daily backup process (System | System | Automatic Backup).
Restore This command is available with IP500 V2 systems. When selected, Manager copies the
System Files folders and files from the System SD card's /backup folder to its /primary folder. Any
matching files and folders already present are overwritten.

Warning:
• After this command is completed, the system is rebooted. This will end all calls and
services in progress.
Upgrade This command is available for IP500 V2 systems that have a system SD card and Optional
Binaries SD card installed.
When this command is selected, all files except config.cfg and keys.txt files in the
Optional SD card's \primary folder are copied to the System SD card.

Warning:
• After this command is completed, the system is rebooted. This will end all calls and
services in progress.
Table continues…

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Command Description
Upgrade This command is available for IP500 V2 systems that have a system SD card and Optional
Configuration SD card installed.
When this command is selected, any config.cfg and keys.txt files in the Optional SD
card's \primary folder are copied to the System SD card.

Warning:
• After this command is completed, the system is rebooted. This will end all calls and
services in progress.
Upload Phone This command is available for IP500 V2 control units. When this command is selected,
Files Manager copies the software files relating to phone firmware to the memory card. For IP500
V2 control units, use Upload System Files.
Copy System This command is available for IP500 V2 systems that have an Optional SD card installed in
Card addition to the mandatory System SD card. When this command is selected, the system will
copy the folders and files on its System SD card to the Optional SD card. Any matching files
and folders already present on the Optional SD card are overwritten.
This process takes at least 90 minutes and can take longer.
Configuration This command will exit Embedded File Management and return Manager to configuration
editing mode.

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Part 3: Security Configuration

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Chapter 15: Security Administration

The security settings for access to an IP Office system are separate from the configuration settings.
You can only view and edit the security settings directly from the IP Office. You cannot save the
security settings as a file on your PC.
This section provides an overview of the main security settings. For more information, see the
Avaya IP Office™ Platform Security Guidelines manual.
You can setup security using the following elements:
• Access control to prevent unauthorized use.
• Encryption to guarantee data remains private.
• Message authentication to ensure that the data has not been tampering with.
• Identity assurance to verify the data source.
Related links
Service Users, Roles, and Rights Groups on page 127
Default Service Users and Rights Groups on page 129
Default Rights Groups on page 130
Access Control on page 132
Encryption on page 133
Message Authentication on page 134
Certificates on page 135
Implementing Security on page 135
SRTP on page 137

Service Users, Roles, and Rights Groups

The IP Office controls access to its settings and services using Service Users and Rights
Groups stored in its security settings.
• Connecting to the IP Office requires entering a Service Users username and password.
• The Rights Groups to which the Service Users belongs define the permissions that the
service user has.

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Security Administration

Feature Description
Security Administrator The security administrator is a special user that differs from the service users. You
can use their username and password to access and edit the security settings.
However, the security administrator cannot access any other IP Office services.
You cannot remove or disable this account.
Service Users Each service user has a username, a password, and is a member of one or more
Rights Groups. The IP Office supports up to 64 service users.
Rights Groups The Rights Groups to which a service user belongs sets their permissions. For
example:
• Set whether the service user can view and/or edit the configuration settings.
• Set which parts of the configuration settings the service user can access.
• Set whether the service user can view and/or edit the security settings.
• Set whether the service user can change their password.
When a service user is a member of more than one rights group, they combine the
permissions of each rights group. The IP Office supports up to 32 rights groups.

Example Rights Assignment

In the example above:

• Service user X can read and write the configuration settings. However, they can only edit
operator settings, and can only make changes that are mergeable.
• Service user Y can read and write the configuration settings. They can edit all the
configuration settings, including making changes that require an IP Office reboot. They can
also access the settings of the Voicemail Pro service.
• Service user Z has the same configuration access as service user Y. However, they can also
view and edit the security settings.
• The security administrator can only view and edit the security settings.
Changing Administrative Users and Rights Groups
You can use IP Office Manager and IP Office Web Manager to edit service users and rights
groups. Before making any changes, you must consider the following:
• IP Office in a multi-site network must have consistent service users and rights groups. IP
Office Manager and IP Office Web Manager have synchronization tools to assist in achieving
this.

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 Default Service Users and Rights Groups

• All changes must follow security best practices. For example, following a password policy
and only allowing minimal access rights.
Related links
Security Administration on page 127

Default Service Users and Rights Groups

The following information is applicable for IP Office R11.1FP2.
Security Administrator Account
This is the default security administration account and has all rights to all security settings. You
cannot remove or disable this account.
Default Service User Accounts
The following service user accounts are present on the first start-up, and after a security settings
reset:
Name Account Description/Default Rights Default Rights Group
Status Membership
Administrator Enabled This service user is the default account for IP Administrator Group
Office configuration. Do not remove, disable, or
System Status
rename this service user.
Group
Business Partner
AdjunctServer Disabled Subscription mode IP Office systems use this Adjunct Server
service user to enable COM support for an IP
Office application server.
BranchAdmin Disabled The IP Office uses this service user for IP SMGR Admin
Office branch systems managed by SMGR.
BuisnessPartner Disabled The IP Office uses this service user for Business Partner
configuration access by business partners.
COMAdmin Enabled Subscription mode IP Office systems using this COM Admin
service user for connection to COM.
DirectoryService Enabled The IP Office uses this service user for HTTP Directory Group
directory access.
EnhTcpaService Enabled The IP Office uses this service user for TCPA Group
connection with the Avaya one-X® Portal
service.
IPDectService Disabled The IP Office uses this service user for DECT IPDECT Group
R4 system provisioning.
Maintainer Disabled The IP Office uses this service user for back Maintainer
up, restore and upgrade connections.
Table continues…

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Security Administration

Name Account Description/Default Rights Default Rights Group

Status Membership
MCMAdmin Disabled The IP Office uses this service user for MCM Admin
connection to Customer Operations Manager.
TURNServer Disabled The IP Office uses this service user fto support TURN Server
User Portal WebRTC users using TURN.

Related links
Security Administration on page 127

Default Rights Groups

The following information is applicable for IP Office R11.1FP2 SP4 and higher. The following rights
groups are present on first start-up and after a security settings reset.
Rights Group Settings
Rights Group Rights Set Rights Enabled
Administrator Configuration IP Office Service All
Group Rights
Manager Operator Administrator
Rights
External IP Office Service Media Manager Administrator, Reporter
Rights Administrator
System Status System Status IP Office Service All
Group Rights
TCPA Group Telephony APIs IP Office Service Enhanced TSPI Access, DevLink3
HTTP Rights Directory Read, Directory Write
IPDECT Group HTTP IP Office Service DECT R4 Provisioning, Directory Read
Rights
SMGR Admin Web Services IP Office Service All except Service Monitor Read
Rights
Web Manager All except Service Change
Rights
Business Partner Configuration IP Office Service All
Security Rights All
Administrator
System Status All
Web Services All except Service Monitor Read
All except Service Change
Table continues…

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 Default Rights Groups

Rights Group Rights Set Rights Enabled

External Web Manager Voicemail Pro Administrator, one-X
Rights Portal Administrator, Web Control
Administrator, WebRTC Gateway
Administrator, Authentication Module
Server Administrator
Maintainer Configuration IP Office Service Read All Configuration
System Status Rights All
Web Services Configuration Read All, Backup,
Restore, Upgrade
External Voicemail Pro Basic, one-X Portal Super
User, Web Control Administrator, Web
Control Security
Directory Group HTTP IP Office Service Directory Read, Directory Write
Rights
COM Admin Web Services IP Office Service Security Write Own Password, Backup,
Rights Restore, Upgrade
MCM Admin Security IP Office Service Write Own Service User Password
Administrator Rights
Web Services Backup, Restore, Upgrade
Adjunct Server External IP Office Service Adjunct Server
Rights
TURN Server External IP Office Service TURN Server Connection
Rights

Additional Rights Groups for Non-Subscription Systems

The IP Office creates these additional default rights groups on non-subscription mode systems.
They have no associated default service users.
Rights Group Rights Set Rights Enabled
Manager Group Configuration IP Office Service All
Rights
Manager Operator Manager
Rights
Operator Group Configuration IP Office Service All
Rights
Manager Operator Operator
Rights
Security Admin Security IP Office Service All
Administrator Rights
Backup Admin Web Services IP Office Service Backup, Restore
Rights
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Security Administration

Rights Group Rights Set Rights Enabled

External IP Office Service one-X Portal Super User
Rights
Upgrade Admin Web Services IP Office Service Upgrade
Rights
System Admin Configuration IP Office Service Read All Configuration, Write All
Rights Configuration, Merge Configuration
Web Services IP Office Service Security Write Own Password,
Rights Configuration Read All, Configuration
Write All
Web Manager All except Service Change
Rights
External IP Office Service Voicemail Pro Standard, one-X
Rights Portal Administrator, WebRTC Gateway
Administrator
Maint Admin Web Services IP Office Service Backup, Restore, Upgrade
Rights
Customer Admin Web Services IP Office Service Security Write Own Password,
Rights Configuration Read All, Configuration
Write All, Backup, Restore, Upgrade
Web Manager All except Service Change
Rights
External IP Office Service Voicemail Pro Standard, one-X Portal
Rights Super User
Management API Web Services IP Office Service Management API Read, Management
Group Rights API Write
TURN Server External IP Office Service TURN Server Connection
Rights

Related links
Security Administration on page 127

Access Control
The IP Office uses service user and rights group settings to control access to the IP Office
settings. All connections to an IP Office service require a service user name and password. That
service user must be a member of a rights group with permission to access the require service
and perform the required actions.

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 Encryption

Example Rights Assignment

In the example above:

• Service user X can read and write the configuration settings. However, they can only edit
operator settings, and can only make changes that are mergeable.
• Service user Y can read and write the configuration settings. They can edit all the
configuration settings, including making changes that require an IP Office reboot. They can
also access the settings of the Voicemail Pro service.
• Service user Z has the same configuration access as service user Y. However, they can also
view and edit the security settings.
• The security administrator can only view and edit the security settings.
Related links
Security Administration on page 127

Encryption
Encryption ensures no one else can read the data sent to and from the IP Office. Encryption is the
application of a complex mathematical process at the originating end, and a reverse process at
the receiving end. The process at each end uses the same ‘key’ to encrypt and decrypt the data:

The IP Office can encrypt any data sent using a number the following algorithms:

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Security Administration

Algorithm Key size (bits) Use

DES-40 40 Not supported.
DES-56 56 Not supported.
3DES 112 Low security.
RC4-128 128 Medium security.
AES-128 128 High security.
AES-256 256 High security.

In general, the larger the key size, the more secure the encryption. However, smaller key sizes
require less processing. The system supports encryption using the Transport Layer Security (TLS)
protocol.
Related links
Security Administration on page 127

Message Authentication
Message authentication enables detection of any alteration to data to and from IP Office. To
support authentication, the originator of the data also sends a signature (called a hash) of the data
sent. The receiver can then check that the data and the signature received match.

The IP Office can authenticate data using the following algorithms:

Algorithm Hash size (bits) Use
MD5 128 Not recommended.
SHA-1 160 ‘Acceptable’ security.
SHA-2 256, 384, 512 ‘Strong’ security

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 Certificates

In general, the larger the hash size, the more secure the signature. However smaller hash sizes
require less processing.
IP Office supports message authentication using the Transport Layer Security (TLS) 1.0, 1.1, and
1.2 protocol.
Related links
Security Administration on page 127

Certificates
Public key cryptography is one of the ways to maintain a trustworthy networking environment.
A public key certificate (also known as a digital certificate or identity certificate) is an electronic
document used to prove ownership of a public key. The certificate includes information about the
key, information about its owner's identity, and the digital signature of an entity that has verified the
certificate's contents are correct. If the signature is valid, and the person examining the certificate
trusts the signer, then they know they can use that key to communicate with its owner.
For more information, see Certificate Management on page 638.
Related links
Security Administration on page 127

Implementing Security
The IP Office has a range of security features. However, for ease of initial IP Office installation
the security features are not enabled by default. Therefore, during installation it is necessary to
implement the configuration options listed here.
Minimum Security
A minimum-security scenario is one where any individual with the correct service user name and
password can access the configuration from any PC using IP Office Manager. Passwords can be
simple and never age.
• Change the default passwords of all service users and the security administrator
• Set the system Security Administration service security level to Secure, Low.
• Leave the system service user Password Reject Action set to Log to Audit Trail.
• Leave the system Client Certificate Checks level set to None.
• Leave the system Minimum Password Complexity set to Low.
• Leave the system Previous Password Limit set to 0.
• Leave the system Password Change Period set to 0.

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Security Administration

• Leave the system Account Idle Time set to 0.

• Leave the Certificate Check Level to Low in the IP Office Manager preferences.
Medium Security
A medium-security scenario uses password complexity restrictions. Passwords cannot be simple
and will age.
• Change the default passwords of all service users and the security administrator
• Set the system Security Administration service security level to Secure, Medium.
• Set the system Configuration service security level to Secure, Medium.
• Leave the system service user Password Reject Action set to Log to Audit Trail.
• Leave the system Client Certificate Checks level set to None.
• Set the system Minimum Password Complexity to Medium.
• Set the system Previous Password Limit to a non-zero value.
• Set the system Password Change Period to non-zero value.
• Set the system Account Idle Time to a non-zero value.
• Disable all the system Unsecured Interfaces.
• Leave the Certificate Check Level to Low in the IP Office Manager preferences.
Maximum Security
A maximum-security scenario is one where both configuration and security settings are
constrained. Certified individuals with the correct service user name and password can access
the configuration from specific PC installations of IP Office Manager. Passwords cannot be simple
and will age. IP Office Manager can manage specific systems.
• Change the default passwords of all service users and the security administrator
• Set the system Security Administration service security level to Secure, High.
• Set the system Configuration service security level to Secure, High.
• Set the system service user Password Reject Action to Log and Disable Account.
• Set the system Client Certificate Checks level to High.
• Set the system Minimum Password Complexity to High.
• Set the system Minimum Password Length to greater than 8.
• Set the system Previous Password Limit to greater than 5.
• Set the system Password Change Period to a non-zero value.
• Set the system Account Idle Time to a non-zero value.
• Install 1024-bit+ third-party certificates in all IP Office server certificates, derived from a
trusted certificate authority.
• Install the corresponding trusted CA certificate in each of the IP Office Manager PC’s
Windows certificate stores.
• Install 1024-bit+ third-party certificates in all IP Office Manager Certificate Stores.

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 SRTP

• Install the corresponding certificates in all the system Certificate Stores of all permissible
Manager entities, and the trusted CA certificate.
• Disable all the system Unsecured Interfaces.
• Set the Manager Certificate Checks level to High in the IP Office Manager preferences.
• Set the certificate offered to the system in the IP Office Manager preferences.
The above essentially locks the IP Office and corresponding IP Office Manager together. Only
recognized (by strong certificate) entities can communicate successfully on the service interfaces.
All services use strong encryption and message authentication.
The use of intermediate CA certificates can overcome the limit of 6 certificates in each system IP
Office certificate store.
Related links
Security Administration on page 127

SRTP
Secure Real-Time Transport Protocol (SRTP) refers to the application of additional encryption and
or authentication to VoIP calls (SIP and H.323). The IP Office can apply SRTP to calls between
phones, between ends of an IP trunk or in various other combinations.
IP Office supports:
• Individual configuration for RTP and RTCP authentication and encryption.
• HMAC SHA1 as the authentication algorithm.
• AES-CM as the encryption algorithm.
• 80-bit or 32-bit authentication tag.
• Key length of 128-bits.
• Salt length of 112-bits.
You can configure the use of SRTP at the system level. The options are Best Effort or Enforced.
The recommended setting is Best Effort. In that scenario, the IP Office uses SRTP if supported
by the other end. When using Enforced, the IP Office does not allow the call if the other end does
not support SRTP.
You can set different SRTP settings for individual trunks and extensions if necessary. The IP Office
supports SRTP on SIP Lines, SM Lines, and IP Office Lines.
Encrypted RTCP
The IP Office supports unencrypted RTCP by default. You can configure encrypted RTCP when
required.
For SRTP calls where one end is using encrypted RTCP and the other is unencrypted, the call
cannot use direct media. Instead, the IP Office provides SRTP relay for the call.

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Security Administration

Authentication
The IP Office supports applying authentication to the voice (RTP) and or control signal (RTCP)
parts of a call. The IP Office applies authentication after applying encryption. That allows
authentication at the remote end before needing to decrypt.
• For the initial exchange of authentication keys during call setup, the IP Office uses SDESC
for SIP calls and H235.8 for H.323 calls.
• The IP Office only supports SRTP when using an addition method such as TLS or a VPN
tunnel to establish a secure data path before call setup.
• A replay attack is when someone intercepts packets and then attempts to use them to for
a denial-of-service or to gain unauthorized access. Replay protection records the sequence
of packets received. All RTP and RTCP packets in the call stream have a sequential index
number. However, the packets can arrive in non- sequential order.
The IP Office protects against replay attacks by using a moving replay window containing the
index numbers of the last 64-authenticated packets received or expected. Using this
- The IP Office only accepts packets that have an index ahead of or inside the replay
window.
The IP Office rejects previously received packets.
• Rekeying is the sending of new authentication keys at intervals during a secure call. The IP
Office does not support rekeying, it sends authentication keys at the start of the call.
Emergency Calls
The IP Office allows emergency calls from an extension regardless of the SRTP requirements and
support.
SRTP Indication
SRTP call indication depends on the model of phone. The System Status Application and
SysMonitor applications can display details of SRTP calls.
Related links
Security Administration on page 127

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Chapter 16: Editing IP Office Security
Settings in Manager

The following conditions apply when editing the IP Office security settings.
• Editing of security settings may only be done online to a system.
No offline saving or editing is allowed for security purposes.
• No errors in the security settings are allowed to persist.
This prevents the system becoming inaccessible through operator error.
• Sets of changes to security objects may be made without the need for the OK button to be
selected every time.
This allows a coordinated set of changes to be accepted or canceled by the operator.

Loading Security Settings

About this task
The address ranges in which Manager searches for systems are set through the Manager
preferences (File | Preferences | Discovery). The security mechanism used for security settings
transfer between Manager and a system are set through the Secure Communications attribute of
Manager preferences (File | Preferences | Security).
If not already done, switch Manager to security mode by selecting File | Advanced | Security
Settings.
Note:
If the system's configuration settings have already been loaded using a service user name
and Password that also has security access, then the security settings are automatically
loaded when Manager is switched to security mode.
Procedure
1. If already in security mode, click in the main toolbar or select File | Open Security
Settings from the menu bar.
2. The Select IP Office window appears, listing those systems that responded.
The list can be sorted by clicking on the column names.

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Editing IP Office Security Settings in Manager

3. If the system required was not found, the address used for the search can be changed.
Enter or select the required address in the Unit/Broadcast Address field and then click
Refresh to perform a new search.
4. When the system required is located, check the box next to the system and click OK.
5. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
6. If the security settings are received successfully, they appear within Manager.
• If the service user name/password is incorrect, or the service user has insufficient rights
to read the security settings, "Access Denied" is displayed.
• If the network link fails, or the secure communication mode is incorrect (for example
Manager is set to unsecured, but the system is set to secure only), "Failed to
communicate with IP Office" is displayed.

Saving Security Settings

Procedure
1. Click in the Main Toolbar or select File | Save Security Settings from the menu bar.
These options are only available when some change has been made.
2. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.

Resetting a System's Security Settings

Procedure
1. Select File | Reset Security Settings (if in security mode), or File | Advanced | Erase
Security Settings (if in configuration mode).
2. The Select IP Office window appears, listing those systems that responded. The list can be
sorted by clicking on the column names.
3. When the system required is located, check the box next to the system and click OK.
4. The user name and password request for the system is then displayed. Enter the required
details and click OK. By default this is a different user name and password from those that
can be used for configuration access.
5. Manager will indicate if the security settings are reset successfully.

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Chapter 17: Security Mode Field
Descriptions

The Manager Security Mode is used to load and edit the security settings of a system. How the
controls operate is similar to Manager in configuration mode.
To switch to Security Mode, select File | Advanced | Security Settings.
To switch back to Configuration Mode, select File | Configuration.
Security Mode Screen Elements
Icon Action
Get the Security Settings.

Save the Security Settings.

Not Used in security mode.

Show/Hide the Navigation Pane.

Show/Hide the Group Pane.

Not used in security mode.

Not used in security mode.

Security Settings Pane

This pane is used to select the type of security records displayed in the group pane or details pane.
Icon Description
General
Defines general security controls for the system. When selected, the settings are displayed in the
details pane.
System
Defines security settings for the system such as application access. When selected, the settings are
displayed in the details pane.
Services
Secure services supported by the system. Currently these are access to security settings and
access to configuration settings.
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Security Mode Field Descriptions

Icon Description
Rights Groups
Create groups with different access rights. When selected, the existing Rights Groups are displayed
in the group pane.
Service Users
Sets the name and password for an administrator. Also allows selection of the Rights Groups to
which the user belongs. When selected, the existing service users are displayed in the group pane.

Group Pane: This pane is used to display the existing Right Groups or Service Users when those
options are selected in the security settings pane.
Details Pane: This pane shows the settings selected in the security settings pane or the group
pane.
Status Bar: This bar display messages about communications between Manager and systems. It
also displays the security level of the communications by the use of a padlock icon.
Related links
General Security Settings on page 142
System on page 146
Services on page 158
Rights Groups on page 160
Service Users on page 167

General Security Settings

Related links
Security Mode Field Descriptions on page 141
General on page 142

General
Security Administrator
The security administrator is a special account that cannot be deleted or disabled. It can be used
to access the system's security settings but cannot access the system's configuration settings.
Field Description
Unique Security Default = Off
Administrator
This setting is no longer used. It is greyed out and set to off, meaning that permission to
access ad change security settings can also be assigned to other service user accounts
through their rights groups memberships.
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 General Security Settings

Field Description
Name Default = 'security'. Range = 6 to 31 characters.
The name for the security administrator.
Change Password Range = 9 to 31 characters.
The password for the security administrator. In order to change the security administrator
password, the current password must be known. The user's original password is set
during the initial configuration of the system.
Minimum Default = Medium.
Password
The password complexity requirements. The options are:
Complexity
• Low - Any password characters may be used without constraint. Password must not
contain your user name.
• Medium - The password must include characters from at least 2 of the character sets
listed below. For example a mix of lower case and upper case. In addition, 3 or more
consecutive identical characters of any type is not allowed.
- Lower case alphabetic characters.
- Upper case alphabetical character.
- Numeric characters.
- Non-alphanumeric characters, for example # or *.
• High - As per medium but requiring characters from at least of the 3 character sets
above.
Previous Password Default = 24. Range = 0 (Off) to 24 records.
Limit (Entries)
The number of previous password to check for duplicates against when changing the
password. When set to 0, no checking of previous passwords takes place. This setting is
active for attempted password changes on both Security Manager and the system.

Phone Registration
Field Description
Block Default IP Default = On
Phone Passcodes
If selected, existing IP phone registrations with default passcodes are not allowed in
the system. Administrators must type in passwords for registering the existing phones.
If not checked, existing IP phone registrations with default passcodes are allowed for
registration with the system. Allowing existing phones to register with default passcodes
pose a security risk as outsiders can access the system using those passcodes.

Service User Details

These settings control service user names and password/account policies. This setting is active
for attempted password changes on all administration interfaces.

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Security Mode Field Descriptions

Field Description
Minimum Name Default = 6, Range 1 to 31 characters.
Length
This field sets the minimum name length for service user names.
Minimum Default = 9, Range 1 to 31 characters.
Password Length
This field sets the minimum password length for service user passwords.
Password Reject Default = 3, Range 0 (Off) to 255.
Limits (Attempts)
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed.
Password Reject Default = Log and Temporary Disable.
Action
The action performed when a user reaches the Password Reject Limit. The options are:
• No Action
• Log to Audit Trail - Creates a record in the system's audit trail indicating the service
user account name and time of last failure.
• Log and Disable - Create an audit trail record and disables the service user account.
The account can only be re-enabled through the service user settings.
• Log and Temporary Disable - Create an audit trail record and temporarily disables the
service user account for 60 seconds.
Minimum Default = Medium.
Password
The password complexity requirements. The options are:
Complexity
• Low - Any password characters may be used without constraint. Password must not
contain your user name.
• Medium - The password must include characters from at least 2 of the character sets
listed below. For example a mix of lower case and upper case. In addition, 3 or more
consecutive identical characters of any type is not allowed.
- Lower case alphabetic characters.
- Upper case alphabetical character.
- Numeric characters.
- Non-alphanumeric characters, for example # or *.
• High - As per medium but requiring characters from at least of the 3 character sets
above.
Previous Password Default = 24. Range = 0 (Off) to 24 records.
Limit (Entries)
The number of previous password to check for duplicates against when changing the
password.
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 General Security Settings

Field Description
Account Password Default = 0 (Off). Range 0 to 999 days.
Change Period
Sets how many days a password is valid following a password change. Note that the
(days)
user must be a member of a rights group that has the option Write own service user
password enabled.
• Whenever this setting is changed, the system recalculates all existing service user
password timers.
• If this timer expires, the service user account is disabled. The account can only be
re-enabled through the service user settings.
• To prompt the user a number of days before the account is locked, set a Expiry
Reminder Time (days) (see below).
Account Idle Time Default = 0 (Off). Range 0 to 999 days.
(days)
Sets how many days a service user account can be inactive before it becomes disabled.
The idle timer is reset whenever a service user successfully logs in.
• If this timer expires, the service user account is disabled. The account can only be
re-enabled through the service user settings.
• Whenever this setting is changed and the OK button is clicked, the system recalculates
all existing service user idle timers.
Expiry Reminder Default = 10. Range 0 (Off) to 999 days.
Time (days)
Sets the period before password or account expiry during which a reminder indication is
shown when the service user logs in. Reminders are sent, for password expiry due to
the Account Password Change Period (days) (above) or due to the individual service
user's Account Expiry date – whichever is the sooner. Currently Manager displays
reminders but System Status does not.

IP Office User Details

These settings control IP Office user password/account policies.
Field Description
Password Default = On.
Enforcement
When enabled, password settings are enforced. When disabled, password requirements
are not enforced and the remaining settings are not editable
Minimum Default = 9, Range 1 to 31 characters.
Password Length
This field sets the minimum password length for user passwords
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Security Mode Field Descriptions

Field Description
Minimum Default = Medium.
Password
The password complexity requirements. The options are:
Complexity
• Low - Any password characters may be used without constraint. Password must not
contain your user name.
• Medium - The password must include characters from at least 2 of the character sets
listed below. For example a mix of lower case and upper case. In addition, 3 or more
consecutive identical characters of any type is not allowed.
- Lower case alphabetic characters.
- Upper case alphabetical character.
- Numeric characters.
- Non-alphanumeric characters, for example # or *.
• High - As per medium but requiring characters from at least of the 3 character sets
above.
Password Reject Default = 5, Range 0 (Off) to 255 failures.
Limits (Attempts)
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed.
Password Reject Default = Log and Temporary Disable.
Action
The action performed when a user reaches the Password Reject Limits (Attempts).
The options are:
• No Action
• Log to Audit Trail - Creates a record indicating the user account name and time of last
failure.
• Log and Disable - Creates an audit trail record and additionally permanently disables
the user account. The account can be enabled using the Account Status field on the
User > User page.
• Log and Temporary Disable - Creates an audit trail record and additionally
temporarily disables the user account for 60 seconds.

Related links
General Security Settings on page 142

System
Related links
Security Mode Field Descriptions on page 141
System Details on page 147
Unsecured Interfaces on page 149
Certificates on page 150

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 System

System Details
Base Configuration
Field Description
Services Base TCP Default = 50804. Range = 49152 to 65526.
Port
This is the base TCP port for services provided by the IP Office. It sets the ports on
which the IP Office listens for requests to access those services, using its LAN1 IP
address. Each service uses a port offset from the base port value.
• If this value is changed from its default, the IP Office Manager application must be
set value through its File > Preferences > Preferences > Services Base TCP Port
setting.
• For information on IP Office port used, see the Using IP Office System Monitor
manual.
Maximum Service Default = 64.
Users
This is a fixed value for information only. The maximum number of service users that
you can configure in the IP Office system's security settings
Maximum Rights Default = 32.
Groups
This is a fixed value for information only. The maximum number of rights groups that
you can configure in the IP Office system's security settings.

System Discovery
System discovery is the processes used by applications to locate and list available systems. If
required, you can disable the IP Office from responding to this process. If you do that, access to
the IP Office requires its specific IP address.
Field Description
TCP Discovery Default = On.
Active
If enabled, the IP Office responds to TCP discovery requests.
UDP Discovery Default = On.
Active
If enabled, the IP Office responds to UDP discovery those requests.

Security
These settings cover the per-system security aspects, primarily TLS settings.
Field Description
Security Session ID Default = 10 hours, Range 0 to 100 hours.
Cache
This sets how long the IP Office system retains TLS session IDs. If retained, the
session ID may be used to quickly restart TLS communications between the system
and a re-connecting application. When set to 0, no caching takes place and each TLS
connection is renegotiated.
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Security Mode Field Descriptions

Field Description
HTTP Challenge Default = 10.
Timeout (sec)
For HTTP/HTTPS connection attempts, this field sets the timeout for connection
validation responses.
RFC2617 Session Default = 10.
Cache (mins)
For HTTP/HTTPS sessions, this field sets the duration for successful logins as per
RFC2617.
Minimum Protocol Default = TLS 1.2
Version
This sets the minimum TLS protocol version for TLS connections.

HTTP Ports
These settings set the ports for web-based configuration access to the system.
Field Description
HTTP Port Default = 80.
HTTPS Port Default = 443.
Web Services Port Default =8443.

Web Socket Proxy

These settings are applicable to WebSocket communication over IP Office lines.
Field Description
Enabled Default = On.
• When enabled, IP Office Web Manager uses the proxy server to communicate
between the Server Edition Primary server and other IP Office nodes.
• When disabled, the WebSocket proxy is disabled. All IP Office line WebSocket
communication is closed with 404 NotFound.
Enforce Secure Default = On.
• When enabled, all proxy communication over IP Office line Websocket uses HTTPS.
• When disabled, all HTTPS IP Office line Websocket communication is closed with
403 Forbidden.

Avaya Spaces Configuration Details

Field Description
Avaya Spaces API The API key and key secret used for connection between the IP Office and Avaya Cloud
Key Services. For further details, see the IP Office Avaya Workplace Client Installation Notes
Avaya Spaces Key
Secret

Related links
System on page 146

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 System

Unsecured Interfaces
These features relate to applications that access the system configuration settings using older
security methods.
Field Description
System Password Range = 0 to 31 characters.
The system password is for the following:
• IP Office Manager access to upgrade IP Office IP500 V2 systems.
• UDP/TCP access by SysMonitor if the Monitor Password password is blank.
Voicemail Password Default = Blank. Range = exactly 31 characters.
For IP Office 11.1 FP1 and higher versions, the password for voicemail connection is
enforced to 31 characters.
• This password is also set through the Voicemail Pro client and Web Manager
application.
• When no password is set, an auto generated password is automatically set on both
Voicemail Pro client and Web Manager systems.
Monitor Password Default = Blank. Range = 0 to 31 characters.
This password is used by SysMonitor for UDP and TCP access. If blank, then
SysMonitor uses the System Password.
If changing this password with no previous password set, enter the system password as
the old password.
Use Service User Default = Off.
Credentials
If enabled, SysMonitor access using UDP or TCP, uses service user names and
passwords rather than the Monitor Password. The service user must also be a
member of a rights group with System Status > > System Monitor - Access enabled.

Application Controls
These check boxes control which actions the system will support for legacy applications. Different
combinations are used by the different applications. A summary of the applications affected by
changes is listed in the Application Support list.
• For Linux-based IP Office servers, some ports, such as port 69 and 80, are also controlled by
the Solution > > Platform View > Settings > System > Firewall Settings.
Field Description
TFTP Server Default = On.
TFTP Directory Default = Off.
Read
Used by DECT R4 for IP Office contacts if using an AIWS.
TFTP Voicemail Default = Off.
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Security Mode Field Descriptions

Field Description
Program Code Default = On.
Controls use of the upgrade wizard from within IP Office Manager.
DevLink Default = On.
Control support for connections from DevLink applications. That includes UDP, TCP and
HTTP access by SysMonitor.
TAPI/DevLink3 Default = Off.
Controls support for connections from TAPI and DevLink3 applications.
HTTP Directory Default = On.
Read
Allows system directory accessed using HTTP rather than HTTPS.
HTTP Directory Default = On.
Write
Allow HTTP rather than HTTPS to import temporary directory records into the system
directory.

Application Support
This panel is shown for information only. It indicates the effect on various applications of the
Application Controls selections.
Related links
System on page 146

Certificates
Additional Configuration Information
For additional information on certificates, see Certificate Management on page 638.
Services between the system and applications can, depending on the settings of the service
being used for the connection, require the exchange of security certificates. The system can either
generate self-signed certificate or use certificates from a trusted source can be loaded.
Identity Certificate
These settings relate to the X.509v3 certificate that the system users to identify itself when
connecting another device using TLS. For example, a PC running IP Office Manager set to
Secure Communications.
The system’s certificate is advertised (used) by services which have their Service Security Level
set to a value other than Unsecure Only.
By default, each IP Office server provides a self-generated certificate, generated when the system
is first installed. However, the certificate can also come from other sources:
• An alternate identity certificate for the system from added using the Set button.
- For secondary, expansion and application servers, this can be an identity certificate
generated for that server from the web control menus of the primary server.

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• For subscription mode systems, Automatic Certificate Management can be selected. COM
then automatically provides the system with an appropriate identity certificate and certificate
updates.
Field Description
Offer Certificate Default = On.
This is a fixed value for indication purposes only. This sets whether the system will offer a
certificate in the TLS exchange.
Offer ID Certificate Default = On
Chain
When enabled, the IP Office advertises a chain of certificates during TLS session
establishment.
• The chain of certificates starts with the system's identity certificate
• It then adds any certificates it finds in its trusted certificate store with the same Common
Name in their "Issued By" Subject Distinguished Name field.
• If the Root CA certificate is found in the trusted certificate store, that is also included in
the certificate chain.
• A maximum of six certificates are supported in the certificate chain.
Issued To Default = IP Office identity certificate.
For information only. The common name of certificate issuer.
Certificate Expiry Default = 60, Range = 30 to 180
Warning Days
IP Office Manager can display a warning when a system’s security certificate is due to
expire. This setting is used to set the trigger for certificate warnings.

The following settings are only shown for subscription mode systems. They allow COM to provide
the system with its identity certificate and to automatically update the certificate when required.
Field Description
Automatic Default = Disabled
Certificate
Supported for subscription mode systems only. When enabled, the system uses an
Management
identity certificate supplied by COM along with a copy of the COM root certificate. The
maintenance and renewal of the identity certificate and its trust chain are performed
automatically.
SAN Details Origin If the identity certificate issued to the system by COM needs to include any location
specific subject alternate name values, this field can be used to define those values.
• Migrate from existing ID certificate - When generating a new certificate for the
system, use the SAN details from its existing identity certificate.
• Generate form current LAN configuration - When generating a new certificate,
create the SAN details from the system's existing LAN and SIP settings.
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Security Mode Field Descriptions

Field Description
Automatic Phone Default = Enabled
Provisioning
This additional option is supported when using Automatic Certificate Management.
When enabled, phone certificates on phones that support certificate download, are
automatically updated when the system identity certificate is updated.
• New and default phones obtain the certificate using the normal trust on first use
process.
• When an update occurs, the 46xxsettings.txt file is updated to includes details of
both certificates. Following a restart, the phones fetch the new certificate using the old
certificate details.

The following settings can be used to manage the current identity certificate.
Field Description
Set Using Set allows you to load an identity certificate and its associated private key.
• This control is not shown for subscription mode systems using Automatic Certificate
Management.
The IP Office supports:
• 1024, 2048 and 4096 bit RSA keys. Use of 4096 RSA keys may impact system
performance.
• SHA-1, SHA-256, SHA-384, and SHA-512 signature algorithms. Using signature size
larger than SHA-256 may impact system performance.
The source may be:
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in the PKCS#12 format.
- Pasted from clipboard in PEM format, including header and footer text. This method
must be used for PEM (.cer) and password protected PEM (.cer) files. The identity
certificate requires both the certificate and private key. The CER format does not
contain the private key. For these file types, select Paste from clipboard and then
copy the certificate text and private key text into the Certificate Text Capture window.
Using a file as the certificate source:
In Manager, when using the file option, the imported file (.p12, .pfx or .cer) can
only contain the private key and identity certificate data. It cannot contain additional
Intermediate CA certificates or the Root CA certificate. The Intermediate CA certificates
or the Root CA certificate must be imported separately into the IP Office Trusted
Certificate Store. This does not apply to Web Manager.

Note:
Web Manager does not accept the file of type CER with extension .cer. That file
type can only be used in Manager.
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 System

Field Description
View Displays details of the current identity certificate. The certificate view menu can also be
used to install the certificate (but not its private key) into the viewing PCs local certificate
store. This can the be used by the PC for secure connection to the system or to export
the certificate from the PC.
Regenerate This command generates a new identity certificate:
• For system's using the system's own self-generated self-signed identity certificate, this
command generates a replacement for the current identity certificate.
• For subscription mode system's, this command requests a replacement identity
certificate from COM. Alternatively, it can be used to request an identity certificate
for another server.

Important:
• Regeneration takes up to a minute, during which time system performance is
impacted. Therefore, only perform this action during a maintenance window. The
regeneration takes places after saving the security settings.
When clicked, the Regenerate Certificate window prompts you to enter the values in the
following table.

Setting Description
Signature Default = SHA256/RSA2048.
Select the signature algorithm and the RSA key length to use for the new self-signed
identity certificate. The options are SHA256/RSA2048 or SHA1/RSA1024.
Subject Name Default = None
Specifies the common name for the subject of this certificate. The subject
is the end-entity or system that owns the certificate (public key). Example:
ipoffice-0123456789AB.avaya.com. If left blank, a system generated subject
name is used.
Subject Alternative Default = None
Name(s)
Specify any Subject Alternative Name (SAN) values to include in the certificate.
• Each entry consists of a prefix, followed by the colon and then the value. Supported
prefixes are DNS, URI, IP, SRV and email.
• Multiple entries can be added, each separated by the comma. The input field has a
maximum size limit of 511 characters.
• Example: DNS:192.168.0.180,IP:192.168.0.18,URI:SIP:example.com
For Different Default = Off
Machine
This option is only shown for subscription mode systems using Automatic Certificate
Management.
When selected, the address details of the other server and the duration of the certificate
(maximum 825 days) are requested. After generating the certificate, the browser
automatically downloads the certificate file.

Certificate Checks

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Field Description
Certificate Expiry Default = 60. Range = 30 to 180 days.
Warning Days
Set the number of days before the expiry of any stored certificate, at which IP Office
Manager, IP Office Web Manager, and System Status Application will display warnings
Use different Default = None
certificate for SIP
The possible settings are None, SIP Trunks or SIP & SM Trunks, SIP Phones.
telephony
• When set to None, all secure telephony communications use the system’s default
identity certificate and settings.
• When set to any other option, an extra set of options similar to those shown for
Identity Certificate section are displayed. These can be used to define the certificate
used for secure telephony communications. The certificate to use is uploaded to the
system’s certificate store using the Set button.
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Field Description
Received Default = None.
certificate checks
This setting is used for HTTPS/TLS administration connections to the system by
(Management
applications such as IP Office Manager when the Service Security Level of the service
interfaces)
being used is set to High.
The received certificate is tested as follows:
• None - The certificate must be in date. No extra checks are made.
• Low - As above but also:
- Check the certificate's public key is 1024 bits or greater..
• Medium - As above, but also:
- Check there is a trust chain from the Trusted Certificate Store (TCS) to the root
Certificate Authority (CA).
- For IP Office R11.1.3 and higher:
• Check that the certificate has a key usage defined.
• If the certificate has extended key usage settings, check they match the purpose
for which the certificate is being used.
• Check that the certificate does not include any unknown extensions marked as
critical.
• Note: For systems upgraded to R11.1.3, these additional checks are only used
after the existing setting is changed. For example, changed from Medium to High
and then back to Medium. It is recommended to backup the configuration before
making any change.
• High - This settings enables implementation of a strict trust domain where only known
certificates are accepted. This is a form of 'certificate pinning' and overcomes the
limitation of the standard tree structure PKI where any certificates issued by the root
CA are always trusted. High uses the same checks as Medium plus:
- Check the certificate's public key is 2048 bits or greater
- Check the certificate is not a self-signed certificate.
- Not reflected.
- Check there is a copy of the certificate in the IP Office system's Trusted Certificate
Store.
• Medium + Remote Checks - Use the same checks as Medium plus the following:
- Perform hostname validation by verifying one of the SAN entries matches the
connection's FQDN. If necessary, the SAN entry used can be an IP address.
- For SIP, verify that the certificate source is authoritative for the SIP domain as per
RFC5922.
• High + Remote Checks - Use the same checks as High plus the same additional
checks as Medium + Remote Checks.
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Field Description
Received certificate Default = None.
checks (Telephony
This setting sets how the IP Office validates the identity certificate it receives for TLS
endpoints)
telephony connections.
• An identity certificate is not installed in all SIP phones. Therefore, for SIP, the IP Office
does not require a client certificate from SIP phones, only from SIP and SM trunks.
The received certificate is tested as follows:
• None - The certificate must be in date. No extra checks are made.
• Low - As above but also:
- Check the certificate's public key is 1024 bits or greater..
• Medium - As above, but also:
- Check there is a trust chain from the Trusted Certificate Store (TCS) to the root
Certificate Authority (CA).
- For IP Office R11.1.3 and higher:
• Check that the certificate has a key usage defined.
• If the certificate has extended key usage settings, check they match the purpose
for which the certificate is being used.
• Check that the certificate does not include any unknown extensions marked as
critical.
• Note: For systems upgraded to R11.1.3, these additional checks are only used
after the existing setting is changed. For example, changed from Medium to High
and then back to Medium. It is recommended to backup the configuration before
making any change.
• High - This settings enables implementation of a strict trust domain where only known
certificates are accepted. This is a form of 'certificate pinning' and overcomes the
limitation of the standard tree structure PKI where any certificates issued by the root
CA are always trusted. High uses the same checks as Medium plus:
- Check the certificate's public key is 2048 bits or greater
- Check the certificate is not a self-signed certificate.
- Not reflected.
- Check there is a copy of the certificate in the IP Office system's Trusted Certificate
Store.
• Medium + Remote Checks - Use the same checks as Medium plus the following:
- Perform hostname validation by verifying one of the SAN entries matches the
connection's FQDN. If necessary, the SAN entry used can be an IP address.
- For SIP, verify that the certificate source is authoritative for the SIP domain as per
RFC5922.
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Field Description
• High + Remote Checks - Use the same checks as High plus the same additional
checks as Medium + Remote Checks.
H.323 Security Level Default = High (Medium for IP500 systems and systems upgrade to R11.1.3 or higher).
Sets the minimum cipher strength the IP Office accepts on TLS connections for H.323
phones and trunks. Not used for clients where ciphers are enabled and chosen based
on those offered by the TLS server.
• This setting replaces the CIPHER_LEVELS_H232 NUSN used by R11.1.2.x systems.
• For further details, see the Avaya IP Office™ Platform Security Guidelines manual.
• Low (0) - Accept low, medium, and high-strength ciphers. Low and medium on IP500
V2 systems.
• Medium (1) - Accept medium and high-strength ciphers. Medium on IP500 V2
systems.
• High (2) - Accept high-strength ciphers. Not supported for IP500 V2 systems.
- For a list of ciphers, see https://documentation.avaya.com/bundle/IPOfficeSecurity/
page/Supported_Ciphers.html.
- High-strength ciphers are GCM ciphers. These are not supported by any model of
IP500 V2 system.
SIP Security Level Default = High (Medium for IP500 V2 systems and systems upgraded to R11.1.3 or
higher).
Sets the minimum cipher strength the IP Office accepts on TLS connections for SIP
phones and trunks. Not used for clients where ciphers are enabled and chosen based
on those offered by the TLS server.
• This setting replaces the CIPHER_LEVELS_SIP NUSN used by R11.1.2.x systems.
• For further details, see the Avaya IP Office™ Platform Security Guidelines manual.
• Low (0) - Accept low, medium, and high-strength ciphers. Low and medium on IP500
V2 systems.
• Medium (1) - Accept medium and high-strength ciphers. Medium on IP500 V2
systems.
• High (2) - Accept high-strength ciphers. Not supported for IP500 V2 systems.
- For a list of ciphers, see https://documentation.avaya.com/bundle/IPOfficeSecurity/
page/Supported_Ciphers.html.
- High-strength ciphers are GCM ciphers. These are not supported by any model of
IP500 V2 system.

Trusted Certificate Store

This section displays a list of the certificates held in the system’s trusted certificate store and
allows management those certificates. Up to 25 X.509v3 certificates can be placed into the store.
When adding a certificate, the source can be:
• Current User Certificate Store.

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Security Mode Field Descriptions

• Local Machine Certificate Store.

• A file in one of the following formats:
- PEM (.cer)
- password protected PEM (.cer)
- DER (.cer)
- password protected DER (.cer)
• Pasted from clipboard in PEM format, including header and footer text.
This method must be used for PKCS#12 (.pfx) files. Select Paste from clipboard and then
copy the certificate text into the Certificate Text Capture window.
SCEP Settings
These settings are used for branch system's which are under centralized management through
SMGR.
Simple Certificate Enrollment Protocol (SCEP) is a protocol intended to ease the issuing of
certificates in a network where numerous devices are using certificates. Rather than having to
individually administer the certificate being used by each device, the devices can be configured to
request a certificate using SCEP.
These settings are normally set during the systems initial configuration.
Field Description
Active Default = Off.
Request Interval Default = 120 seconds. Range = 5 to 3600 seconds.
(sec)
SCEP Server IP Default = Blank.
Address/Name
SCEP Server Port Default = 80 for HTTP and 443 for HTTPS.
SCEP URI Default = /ejbca/publicweb/apply/scep/pkiclient.exe
SCEP Password Default = Blank.

Related links
System on page 146

Services
This tab shows details of the services that the system runs to which service users can
communicate.
Field Description
Name The name of the service. This is a fixed value for information only.
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Field Description
Host System The IP Office system name.
Service Port This is the port on which the IP Office system listens for attempts to access the service.
The routing of traffic to this port must be enabled on firewalls and network devices
between the service users and the IP Office system.
The base port (TCP or HTTP) for each service is offset by a fixed amount from the
ports set in System Settings. For information on port usage, see the IP Office Port Matrix
document on the Avaya support site.
Service Security Sets the minimum security level the service supports.
Level
• If the IP Office system does not already have an X509 security certificate, selecting a
setting other than Unsecure Only will cause the IP Office system to stop responding
for up to a minute whilst it generates a self-signed security certificate.
The options are:
• Unsecure Only - This option allows only unsecured access to the service. The
service's secure TCP port, if any, is disabled. This or disabled are the only options
supported for the System Status Interface and Enhanced TSPI services.
• Unsecure + Secure This option allows both unsecured and secure (Low) access. In
addition, TLS connections are accepted without encryption, just authentication.
• Secure Low - This option allows secure access to the service using TLS and weak (for
example DES_40+MD5) encryption and authentication or higher.
• Secure Medium - This option allows secure access to the service using TLS and
moderate (for example SHA-256) encryption and authentication or higher.
• Secure High - This option allows secure access to the service using TLS and strong
encryption (for example SHA-256) and authentication, or higher.
- Only supported by Linux-based IP Office systems.
- A certificate is required from the client. For IP Office Manager, the Certificates >
Received certificate checks (Management interfaces) setting sets the certificate
checks it uses.
• Disabled - This option is only available for the System Status Interface and Enhanced
TSPI services. If selected, access to the service is disabled.
For details of the ciphers supported by Secure Medium and Secure High, see the
Avaya IP Office™ Platform Security Guidelines manual.
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Field Description
Service Access Used for the Configuration service. Sets the supported modes for IP Office Manager
Source access to the IP Officesystem:
• Server Edition Manager - If selected, the IP Office system can only be configured
using IP Office Manager in its Server Edition mode. This is the default for Server
Edition systems.
- Opening the configuration of a Server Edition system in IP Office Manager running
in any mode other than Server Edition mode should be avoided unless absolutely
necessary for system recovery. Even in that case, IP Office Manager will not allow
renumbering, changes to the voicemail type, and changes to H.323 lines.
• Avaya Aura System Manager - If selected, the IP Office system can only be
configured using SMGR in Branch Mode. This is the default for centrally managed
systems.
• Unrestricted - The IP Office system can be configured using IP Office Manager in its
normal simplified and advanced view modes.

Default Settings
Name Service Port Service Security Level Service Access Source
Configuration 50805 Secure Medium Unrestricted
Security Admin 50813 Secure Medium –
System Status Interface 50809 Secure Medium –
Enhanced TSPI Access 50814 Secure Medium –
HTTP 80, 443 Secure Medium –
Web Services 8443 Secure Medium –
External 50821 Disabled –

Related links
Security Mode Field Descriptions on page 141

Rights Groups
A rights group is a set of permissions to access various features and services. The rights groups
to which a service user belongs sets what that service user can do. If the service user is a
member of several rights groups, they gain the combined permissions of both rights groups.
Related links
Security Mode Field Descriptions on page 141
Group Details on page 161
Configuration on page 161
Security Administrator on page 162
System Status on page 163

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 Rights Groups

Telephony APIs on page 163

HTTP on page 164
Web Services on page 164
External on page 166

Group Details
This tab sets the name of the Rights Group.
Field Description
Name Range = Up to 31 characters
The name for the Rights Group should be unique. The maximum number of rights groups
is 32.

Related links
Rights Groups on page 160

Configuration
This tab sets the configuration settings access for service user's who are members of this Rights
Group.
IP Office Service Rights
Field
Read All If selected, rights group members can read the system configuration.
Configuration
Write All If selected, rights group members can make changes to the system configuration.
Configuration
Merge If selected, rights group members can save configuration changes using a merge.
Configuration
Default If selected, rights group members can default the system configuration.
Configuration
Reboot/Shutdown If selected rights group members can reboot and shutdown the system.
Immediately
Reboot When Free If selected, rights group members can select reboot when free when rebooting the
system.
Reboot At Time Of If selected, rights group members can select reboot at a specific time when rebooting the
Day system.

Manager Operator Rights

This setting controls what types of configuration records Manager will allow members of the
Rights Group to viewed and what actions they can perform with those types of records. The
Administrator and Manager rights group members are also able to access embedded file
management.

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Security Mode Field Descriptions

Role Actions Configuration Record Types

Administrator All View, edit create and delete all configuration records.
Manager View View all except WAN Port.
Edit Extension, User, Hunt Group, Short Code, Service, RAS,
New Incoming Call Route, Directory, Time Profile, Firewall Profile, IP
Route, Least Cost Route, Account Code, ARS.
Delete As edit except Short Code.
Operator View View all except WAN Port.
Edit Extension, User, Hunt Group, Short Code, Service, RAS,
Incoming Call Route, Time Profile, Firewall Profile, IP Route,
Least Cost Route, Account Code, License, ARS.
New None.
Delete Delete Incoming Call Route and Directory.
User & Group Edit View User and Hunt Group records only.
Edit
New None
Delete
User & Group All User and Hunt Group records only.
Administrator
Directory All Directory and Account Code records only.
& Account
Administrator
Time & Attendant All Time Profile and Auto Attendant records only.
Administrator
ICR & User Rights All Incoming Call Route and User Rights records only.
Administrator
Read All View View all configuration records.
Configuration Edit None.
New
Delete

Related links
Rights Groups on page 160

Security Administrator
This tab sets the security settings access for Service user's who are members of this Rights
Group. These settings are ignored and greyed out if a Unique Security Administrator has been
enabled in General Settings.

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Field Description
Read All Security Members of the Rights Group can view the system's security settings.
Settings
Write All Security Members of the Rights Group can edit and return changes to the system's security
Settings settings.
Reset All Security If selected, members of the Rights Group can reset the security settings to default values.
Settings
Write Own If selected, members of the Rights Group can change their own password when requested
Service User to do so by the system. That request may be the result of the Force new password
Password or Account Password Change Period (days) settings. The new password change is
requested automatically at login time.

Related links
Rights Groups on page 160

System Status
This tab sets whether members of the group can access the system using the System Status
Application (SSA).
Field Description
System Status - If selected, members of the Rights Group can view the system's current status and
Access resources using the System Status Application (SSA).
Read All The System Status application includes tools to take a snapshot of the system for
Configuration use by Avaya for diagnostics. That snapshot can include a full copy of the system's
configuration settings. This setting must be enabled for the SSA user to include a copy of
the configuration in the snapshot.
System Control If enabled, the SSA user is able to use SSA to initiate system shutdowns and memory
card shutdown/restarts.
System Monitor - If enabled, members of the Rights Group can use the System Monitor application to
Access perform detailed diagnosis of system problems.

Related links
Rights Groups on page 160

Telephony APIs
Field Description
Enhanced TSPI If selected, applications in this rights group are able to use the system's Enhanced TSPI
Access interface. This interface is currently used by the one-X Portal application server for its
connection to the system.
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Field Description
DevLink3 If selected, applications in this rights group are able to use the system's DevLink3
interface.
This is a TCP based interface that streams real time call events (Delta3 records) and
is the recommended replacement to the existing DevLink Windows based DLL. A new
Rights Group with a user name and password is required for external applications to
connect via the DevLink3 interface.
Location API If selected, applications in this rights group are able to use the system's Location API
interface.

Related links
Rights Groups on page 160

HTTP
This tab sets the HTTP services supported for members of the group.
Field Description
DECT R4 This service is used to allow the system to configure the DECT R4 master base station
Provisioning and to respond to handsets subscribing to the DECT R4 system. It requires both the
system and DECT R4 master base station to be configured to enable provisioning. For full
details, refer to the IP Office DECT R4 Installation manual.
Directory Read If selected, members of the Rights groups have HTTP service read access to directory
records.
Directory Write If selected, members of the Rights groups have HTTP service read and write access to
directory records.

Related links
Rights Groups on page 160

Web Services
These settings are used by users in rights groups using web services to configure and manage
the system. These are currently not used on Standard Mode systems
IP Office Service Rights
Field Description
Security Read All If selected, the rights group members can view system security settings.
Security Write All If selected, the rights group members can change system security settings.
Security Write If selected, members of the Rights Group can change their own password when
Own Password requested to do so by the system. That request may be the result of the Force new
password or Account Password Change Period (days) settings. The new password
change is requested automatically at login time.
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 Rights Groups

Field Description
Configuration If selected, the rights group members can view system configuration settings
Read All
Configuration If selected, the rights group members can change system configuration settings.
Write All
Backup If selected, the rights group members can initiate the system backup process.
Restore If selected, the rights group members can initiate the system restore process.
Upgrade If selected, the rights group members can initiate the system upgrade process.

Web Manager Rights

Field Description
File Manager If selected, the rights group members are assigned Read only access to embedded file
management configuration settings in Web Manager by default.
Service If selected, the rights group members are assigned Read only access to Service
Commands Commands configuration settings in Web Manager by default.
Users, Extensions If selected, the rights group members are assigned Read only access to Users and
Extension configuration settings in Web Manager by default.
Groups, Auto If selected, the rights group members are assigned Read only access to Groups, Auto
Attendant Attendant configuration settings in Web Manager by default.
Incoming Call If selected, the rights group members are assigned Read only access to Incoming
Routes, Alternate Call Routes, Alternate Route Selection, and Short Codes configuration settings in Web
Route Selection, Manage by default.
Short Codes
System, If selected, the rights group members are assigned Read only access to on System,
Locations, Time Locations, Time Profiles, and Licensing configuration settings in Web Manager by default.
Profiles and
Licensing
Lines If selected, the rights group members are assigned Read only access to Lines
configuration settings in Web Manager by default.
Directory, If selected, the rights group members are assigned read only access to Directory,
Authorization Authorization Codes, and Account Codes configuration settings in Web Manager by
Codes, Account default.
Codes
IP Routes, WAN If selected, the rights group members are assigned read only access to IP Routes, WAN
Ports, Firewall Ports, Firewall Profiles, RAS Services Users and Extension configuration settings in Web
Profiles, RAS, Manager by default.
Services, Tunnel
User Rights If selected, the rights group members can access to user rights configuration settings in
Web Manager.

Related links
Rights Groups on page 160

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Security Mode Field Descriptions

External
IP Office Service Rights
These settings are used by users in rights groups for external components using web services to
configure and manage the system.
Field Description
Voicemail Pro If selected, the rights group members can read the configuration and perform backup,
Basic restore, and upgrade.
Voicemail Pro If selected, the rights group members can update the configuration and perform backup,
Standard restore, and upgrade.
Voicemail Pro If selected, the rights group members can update the configuration and security settings.
Administrator
one-X Portal If selected, the rights group members can update the configuration and security settings.
Administrator Does not include backup and restore.
one-X Portal If selected, the rights group members can perform backup and restore.
Super User
Web Control If selected, the rights group members can update the configuration settings.
Administrator
Web Control If selected, the rights group members can update the security settings.
Security
WebRTC Gateway If selected, the rights group members can update the configuration settings.
Administrator
Management API If selected, support the use of the management API to access system configuration
Read settings.
Management API If selected, support the use of the management API to change system configuration
Write settings.
Media Manager If selected, the rights group members can update Media Manager configurations and
Administrator settings. The rights group members can also access all archived recordings.
Media Manager If selected, the rights group members can have read-only access to Media Manager
Standard configurations and access to the recordings.
Reporter If selected, the rights group members can have configuration access to Integrated Contact
Administrator Reporter.
one-X CTI API If selected, support use of one-X CTI API commands.
Adjunct Server Used to support a websocket connection between an IP Office system and an IP Office
Connection application server supporting that system.
TURN Server Allow the name and password details of the rights group's associated service user to be
Connection sent to IP Office User Portal sessions. They then use those details to connect to the
TURN server specified in System | LAN | Network Topology.

Related links
Rights Groups on page 160

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 Service Users

Service Users
These settings are displayed when Service Users is selected in the navigation pane and a
particular service user is selected in the group pane.
The maximum number of service users is 64.
Note that the requirements for these setting (length and complexity) are set through the Service
User Details on the General security settings tab.
Field Description
Name Range = Up to 31 characters.
Sets the service user's name.
• If changing the user name and/or password of the current service user used to load the
security settings, after saving the changes close the configuration.
Password Range = 9 to 31 characters.
Sets the service user's password. Note that when changing a password, a error is
indicated if the password does not meet the service user password rules.
Clear Cache Clears the cache of previous passwords stored when Previous Password Limit (Entries)
is enabled. Allows a previous password to be used again.
Account Status Default = See Default Service Users and Rights Groups on page 129.
Sets whether the account is Enabled, Disabled or Force new password.
• The Password Reject Action on the General security settings tab can automatically
disable an account after too many failed password attempts.
• If an Account Expiration date is set, the account is automatically disabled after that
date.
• A service user set to Force new password if required to set a new password when
logging in. After they enter a new password entered, the account status changes to
Enabled.
Account Default = <None> (No Expiry).
Expiration
You can use this option to set a calendar date after which the account is disabled.
• To prompt the user for a new password before the expiry date, set an Expiry Reminder
Time (days) on the General security settings tab.
Rights Groups Default = See Default Service Users and Rights Groups on page 129.
The check boxes are used to set the rights groups to which the service user account
belongs. The service user's rights will be a combination of all the rights of those groups.

Related links
Security Mode Field Descriptions on page 141

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Part 4: Editing Configuration
Settings

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Chapter 18: Editing Configuration Settings

Related links
IP500 V2 Configuration Operation on page 169
Mergeable Settings on page 171
Configuration Size on page 175
Setting the Discovery Addresses on page 176
Opening a Configuration from a System on page 177
Opening a Configuration Stored on PC on page 180
Creating an Offline Configuration on page 180
Copying and Pasting on page 182
Saving a Configuration onto PC on page 182
Sending an Individual Configuration on page 183
Sending Multiple Configurations on page 184
Erasing the Configuration on page 185
Default Settings on page 186

IP500 V2 Configuration Operation

Before editing the system's configuration settings, it is important to understand how those settings
are stored and used by the system.
The control unit holds copies of its configuration in both its internal non-volatile and RAM memory.
A copy is also held on the System SD card (IP500 V2).
The copies in non-volatile memory and System SD card, are retained even if power to the control
unit is removed. During power up, the system loads the configuration file stored on the System
SD card into its RAM memory. Other systems load the configuration stored in non-volatile memory
into RAM memory. The copy in RAM memory is then used to control the system's operation.
If the system encounters a problem using the configuration file in its System SD card's /primary
folder, it attempt to use the copy in its non-volatile memory. For fully details of the IP500 V2 boot
process and SD card usage refer to the IP Office Installation Manual.
Users actions such as changing their forward destinations or mailbox passcode are written to the
configuration in RAM memory.

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Editing Configuration Settings

Changes made using Manager are written to the configuration in non-volatile memory and then
copied into the RAM memory and System SD.
Between 00:00 and 00:30, a daily backup occurs which copies the configuration in the system's
operation RAM memory back into its non-volatile memory and, on IP500 V2 system's, the System
SD card.On IP500 V2 system, the contents of the system memory cards /primary folder
can then also be automatically copied to the /backup folder by enabling System | System |
Automatic Backup.
When the system is shutdown using the correct shutdown method, the configuration in RAM
memory is copied to the non-volatile memory and System SD card.

Using Manager
When using Manager to edit the configuration settings, the following need to be remembered:
• Manager receives the current configuration settings from RAM memory. Therefore the
configuration it receives includes any changes made by users up to that time. However it
will not contain any subsequent changes made by users.
• When sending the configuration settings back to the system, Manager allows two choices,
reboot or merge.
• Reboot sends the configuration to the system's non-volatile memory along with an instruction
to reboot. Following the reboot, the new configuration in non-volatile memory is copied to the
RAM memory and used.
• Merge sends the configuration to the system's non-volatile memory without rebooting. The
system then copies those changes that are mergeable into the RAM memory. A key point
here is that not all configuration settings are mergeable.
As a result of the above, it is important to bear the follow scenarios in mind:
• Changes made by users after a configuration is received by Manager may be lost when the
configuration is sent back from Manager. Therefore it is preferable to always edit a recently
received copy of the configuration rather than one that has been open for a period of time.

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 Mergeable Settings

• If a merge is attempted with non-mergeable changes, those items will be written to the
non-volatile memory but will not be copied to RAM memory. If a daily backup occurs, they will
then be overwritten by the RAM. If a power loss reboot occurs, they will be written to RAM
memory.
Related links
Editing Configuration Settings on page 169

Mergeable Settings
The menu shown when sending a configuration to the system automatically indicates if the
configuration is mergeable. The table below lists the configuration records that require a system
reboot.
System Settings
Configuration Setting Notes
System Mergeable except Locale and Favor RIP Routes over Static Routes.
LAN | LAN Settings Not mergeable
LAN | VoIP Not mergeable except for:
• Auto-create Extn
• Auto-create User
• H.323 Signaling over TLS
• Remote Call Signaling Port
• Enable RTCP Monitoring on Port 5005
• RTCP collector IP address for phones
• Scope
• Initial keepalives
• Periodic timeout
• VLAN
• 1100 Voice VLAN Site Specific Option Number (SSON)
• 1100 Voice VLAN IDs
The remaining settings are not mergeable. Changes to these settings will require a
reboot of the system.
LAN | Network Topology Not mergeable
Table continues…

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Editing Configuration Settings

Configuration Setting Notes

LAN | DHCP Pools These settings are mergeable. However, the following actions require a merge with
service disruption.
• Changing the Start Address, Subnet Mask or Default Router value for an
existing DHCP Pool of addresses.
• Decreasing Pool Size for an existing DHCP Pool of addresses.
• Deleting an existing DHCP Pool of addresses.
When these actions are performed, the DHCP (Server or Dial In) is re-initialized
which triggers a reboot of the Avaya DHCP Clients (H.323 and SIP) to force
those clients to renew their IP address leases and apply the new settings. For
the remaining Avaya and non-Avaya DHCP clients, you must manually reboot the
devices to force an IP Addresses lease renewal. Otherwise, the devices continue
to use their existing IP addresses until the IP addresses lease expires. IP address
lease time out is set to three days.
DNS Not mergeable
Voicemail Mergeable except Voicemail Type and Voicemail IP Address.
Telephony | Telephony Changes to Companding LAW and Media Connection Preservation require a
reboot.
Telephony | Tones and Changes to the following are not mergeable:
Music
• Disconnect Tone.
• Busy Tone Detection.
• Deleting any Alternate Sources.
System Events | Not mergeable
Configuration
System Events | Alarms Not mergeable
SMTP Not mergeable
VoIP Not mergeable
Dialer Changes to the Operation field or to the Trunk Range / IP Office table require a
reboot.

Line Settings
Configuration Setting Notes
Analog Line | Line These settings are mergeable except Network Type setting.
Settings
Analog Line | Analog The Allow Analog Trunk to Trunk Connect setting is mergeable. The remaining
Options settings are not mergeable.
Table continues…

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 Mergeable Settings

Configuration Setting Notes

BRI Line The following settings are not mergeable.
• Line Sub Type
• Network Type
• TEI
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement
• Clock Quality
• Force Number Plan to ISDN
Decreasing the Number of Channels setting requires a “merge with service
disruption”.
E1 PRI Line The following settings are not mergeable.
• Line Sub Type
• Network Type
• TEI
• Channel Allocation
• CRC Checking
• Clock Quality
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement
• Force Number Plan to ISDN
• Line Signalling
Decreasing the Number of Channels setting requires a “merge with service
disruption”.
E1 PRI Channels The following settings are mergeable: Line Appearance ID (ETSI, ETSI CHI),
Admin (ETSI CHI), Incoming Group ID (ETSI CHI) and Outgoing Group ID
(ETSI CHI). The remaining settings are not mergeable.
E1–R2 Options Changing the Admin setting is mergeable. The remaining settings are not
mergeable.
E1–R2 Channels The following settings are mergeable: Incoming Group ID, Outgoing Group ID
and Admin. The remaining settings are not mergeable.
E1–R2 MFC Group Not mergeable
E1–R2 Advanced Not mergeable
US T1 Line The following settings are mergeable: Admin and Prefix. The remaining settings
are not mergeable.
T1 Channels The following settings are mergeable: Incoming Group ID, Outgoing Group ID,
Line Appearance ID and Admin. The remaining settings are not mergeable.
Table continues…

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Editing Configuration Settings

Configuration Setting Notes

T1 ISDN Line The following settings are mergeable: Prefix, Send Redirecting Number, Admin,
Send original calling party for forwarded and twinning calls and Originator
number for forwarded and twinning calls. The remaining settings are not
mergeable.
T1 ISDN Channels The following settings are mergeable: Incoming Group ID, Outgoing Group ID,
Line Appearance ID, and Admin. The remaining settings are not mergeable.
T1 ISDN TNS Not mergeable
T1 ISDN Special Not mergeable
T1 ISDN Call by Call Not mergeable
SIP Line | SIP Line Mergeable except for the Line Number setting which requires a “merge with
service disruption”.
SIP Line | Transport Changing these settings requires a “merge with service disruption”.
The ITSP Proxy Address and Calls Route via Registrar settings are mergeable.
SIP Line | SIP URI Mergeable except for the Registration and Identity Header settings which
requires a “merge with service disruption”.
SIP Line | SIP Advanced Mergeable except for the Media Connection Preservation setting which requires
a “merge with service disruption”.
Legacy SIP DECT Line | Not mergeable
SIP DECT Base
Legacy SIP DECT Line | Not mergeable
VoIP
SM Line | Session Changing the In Service setting to Disabled (out of service) requires a system
Manager reboot. However, changing the In Service setting to Enabled is mergeable.
Configuration changes made while the line is out of service are also mergeable.
IP Office Line | Short For line subtypes ETSI, ETSI CHI, QSIG A, and QSIG B, these settings are
Codes mergeable.
IP Office Line | VoIP For line subtypes ETSI, ETSI CHI, QSIG A, and QSIG B, these settings are
Settings mergeable.

Extension Settings
Configuration Setting Notes
Extn Mergeable except Base Extension, Extension ID, and Caller Display Type.
Analog Extension | Not mergeable
Analog
H323 Extension | VoIP Not mergeable
SIP Extension | VoIP Not mergeable
IP DECT Extension Mergeable except Reserve License.
SIP DECT Base Not mergeable

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 Configuration Size

WAN Port Settings

Configuration Setting Notes
WAN Port Not mergeable
Frame Relay Not mergeable
DLCIs Not mergeable
Advanced Not mergeable

Tunnel Settings
Configuration Setting Notes
Tunnel (L2TP) Not mergeable
Main (IPSec) Not mergeable
IKE Policies (IPSec) Not mergeable
IKE Policies (IPSec) Not mergeable

Other Settings
Configuration Setting Notes
Control Unit | Unit Not mergeable
License | Remote Server The Reserved Licenses setting is mergeable. The remaining settings require a
reboot.

Related links
Editing Configuration Settings on page 169

Configuration Size
The maximum size of the configuration file that can be loaded into an IP500 V2 control unit is
2.0 MB. When you attempt to save a configuration that is too large, you are warned and the save
canceled.
During normal operation, additional configuration records can be added to the configuration
without using Manager (for example directory records made from phones). If, during the overnight
backup to flash memory, the configuration if found to be too large, records are removed until the
configuration is sufficiently small to be backed up. The records removed are system directory
records and then personal directory in that order.
Note that those records will still exist in the configuration running the system in its RAM memory,
however if the system is restarted they will disappear as the configuration is reloaded from the
Flash memory.
Related links
Editing Configuration Settings on page 169

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Editing Configuration Settings

Setting the Discovery Addresses

By default, when or File | Open configuration is selected, Manager's Select IP Office window
opens. It performs a UDP broadcast to the address 255.255.255.255. This broadcast will only
locate systems that are on the same network subnet as the PC running Manager.
For systems not located on the same subnet as the Manager PC, the following options are
supported.
Description
Specific The Unit/Broadcast Address shown on the Select IP Office menu can be changed to
Addressing the specific IP address of the required system. A single address is routable and so can
be used to discover a system on another subnet.
TCP Discovery A set of TCP addresses and address ranges can be specified for use by the Select IP
Address Ranges Office discovery process.
Known System Manager can write the details of systems it discovers to a file. The list of systems in that
Discovery file can then be used for access to those systems.
DNS Lookup Manager can be configured to locate systems using DNS name lookup. This requires the
systems on a customer network to be added as names on the customer's DNS server
and the Manager PC to be configured to use that server for DNS name resolution. The
use of DNS is configured through File > Preferences > Discovery.

Changing the Initial Discovery Settings

The Discovery tab of the Preferences menu can be used to set the UDP and TCP addresses
used by the discovery process run by the Select IP Office menu.
1. Select File | Preferences menu.
2. Select the Discovery tab.
3. Adjust the settings as required:
Option Description
TCP Discovery Default = On.
This setting controls whether Manager uses TCP to discover systems. The addresses
used for TCP discovery are set through the IP Search Criteria field below.
NIC IP/NIC Subnet This area is for information only. It shows the IP address settings of the LAN network
interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to add
the address range it is part of to the IP Search Criteria. Note that if the address of any of
the Manager PC's NIC cards is changed, the Manager application should be closed and
restarted.
IP Search Criteria This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes, for
example 135.64.180.170 - 135.64.180.175.
Table continues…

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 Opening a Configuration from a System

Option Description
UDP Discovery Default = On
This settings controls whether Manager uses UDP to discover systems.
Enter Broadcast IP Default = 255.255.255.255
Address
The broadcast IP address range that Manager should use during UDP discovery. Since
UDP broadcast is not routable, it will not locate systems that are on different subnets
from the Manager PC unless a specific address is entered.
Use DNS Selecting this option allows Manager to use DNS name (or IP address) lookup to locate
a system. Note that this overrides the use of the TCP Discovery and UDP Discovery
options above. This option requires the system IP address to be assigned as a name on
the users DNS server. When selected, the Unit/Discovery Address field on the Select
IP Office dialogue is replaced by a Enter Unit DNS Name or IP Address field.
SCN Discovery If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as a
single configuration. At least one of the systems in the Small Community Network
must be running Release 6.0 or higher software. See Configuring Small Community
Networking on page 809.
This does not override the need for each system in the Small Community Network to
also be reachable by the TCP Discovery and or UDP Discovery settings above and
accessible by the router settings at the Manager location.

Related links
Editing Configuration Settings on page 169

Opening a Configuration from a System

The initial IP address ranges in which Manager searches for systems are set through the Manager
preferences (File > Preferences > Discovery). By default it scans the local network of the
Manager PC.
Start Manager. If Manager is already started and a configuration is open in it, that configuration
must be closed first.
• If Manager is set to Auto Connect on start up, it will scan for systems automatically and
either display the list of systems discovered or automatically start login to the only system
discovered.

• Otherwise, click on or select File | Open Configuration.

The Select IP Office window appears, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the
configuration of those systems cannot be opened using Manager in Advanced View mode
and the configuration of a Primary Server can only be opened if the Open with Server Edition
Manager option is also selected.

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Editing Configuration Settings

• If Manager has been set with SCN Discovery enabled, systems in a Small Community
Network are grouped together. The checkbox next to the network name can be used to
load the configurations of all the configurations into Small Community Network management
mode.
• If the system required was not found, the Unit/Broadcast Address used for the search
can be changed. Either enter an address or use the drop-down to select a previously used
address. Then click Refresh to perform a new search.
• The address ranges used by Manager for searching can be configured through the File |
Preferences | Discovery tab.
• A list of known systems can be stored and used.
• Manager can be configured to search using DNS names. See the setting File >
Preferences > Discovery > Use DNS.
• Systems found but not supported by the version of Manager being used are listed as Not
Supported.

• If the system detected is running software other than from its primary folder, a warning
icon will be shown next to it. The configuration can still be opened but only as a read-only file.
When you have located the system required, check the box next to the system and click OK.
If the system selected is a Server Editionsystem and Manager is not running in Server Edition
mode, an Open with Server Edition Manager checkbox is shown and pre-selected. Clicking OK
will switch Manager to its Server Edition mode before loading the configuration.
The system name and password request is displayed. Enter the required details and click OK.
The name and password used must match a service user account configured within the system's
security settings.
Additional messages will inform you about the success or failure of opening the configuration from
the system.
The method of connection, secure or insecure, attempted by Manager is set the applications
Secure Communications preferences setting.

• When Secure Communications is set to On, a padlock icon is displayed at all times in the
lower right Manager status field.
• New installations of Manager default to having Secure Communications enabled. This
means Manager by default attempts to use secure communications when opening a
configuration.
• For Server Edition systems, Manager will always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager will
offer to fallback to using unsecured communications.

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 Opening a Configuration from a System

Login Messages
While attempting to login to a system, various additional messages may be displayed.
• Access Denied - This is displayed as the cause if the service user name/password were
incorrect, or the service user has insufficient rights to read the configuration. The Retry
option can be used to log in again but multiple rejections in a 10 minute period may trigger
events, such as locking the user account, set by the Password Reject Limit and Password
Reject Action options in the system's security settings.
• Failed to communicate with system - This is displayed as the cause if the network
link fails, or the secure communication mode is incorrect (for example Manager is set to
unsecured, but the system is set to secure only).
• Account Locked - The account of the service user name and password being used is
locked. This can be caused by a number of actions, for example too many incorrect
password attempts, passing a fixed expiry date, etc. The account lock may be temporary (10
minutes) or permanent until manually unlocked. An account can be enabled again through
the system's security settings.
• Your service user account will expire in X days - This message indicates that an Account
Expiry date has been set on the system service user account and that date is approaching.
Someone with access to the system's security settings will be required to set a new expiry
date.
• Your password will expire in X days. Do you wish to change it now? - This message
indicates that password ageing has been configured in the system's security settings. If your
password expires, someone with access to the system's security settings will be required to
unlock the account.
• Change password - Through the system's security settings, a service user account can be
required to change their password when logging in. The menu provides fields for entering the
old password and new password.
• Retain | Replace | Cancel - This message appears when it is detected that the configuration
of one of the systems in a Server Editionnetwork has previously been edited directly rather
than via access to the primary system.
- Select Replace to replace the system’s update configuration with the copy already held by
the primary server.
- Select Retain to keep the already updated configuration.
- Cancel Select this option to close the configuration without making any changes.
• Contact Information Check - This configuration is under special control - This message
will appear if a Manager user with administrator rights has entered their contact information
into the configuration. For example to indicate that they do not want the configuration altered
while a possible problem is being diagnosed. The options available are:
- Set configuration alteration flag - Select this option if the configuration is being opened
because some urgent maintenance action. When the configuration is next opened, the fact
that it has been altered will be indicated on the System | System tab.
- Delete Contact Information - Select this option to take the system out of special control.
- Leave contact information and flags unchanged - This option is only available to
service users logging in with administrator rights.

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Editing Configuration Settings

Related links
Editing Configuration Settings on page 169

Opening a Configuration Stored on PC

About this task
A configuration file previously saved on the PC can be reopened in Manager. This method of
access does not require entry of a service user name and password. All parts of the configuration
are visible.
Use either of the following processes to load a saved configuration file:
Procedure
1. Click the main toolbar or select File | Offline | Open File from the menu bar. If the files
is one that has previously been opened offline, click the symbol next to in the main
toolbar
2. An Open configuration file window appears.
Use this to browse to the required configuration file.
3. Select the file and click Open.
Related links
Editing Configuration Settings on page 169

Creating an Offline Configuration

About this task
Manager can be used to create a new configuration without connecting to a system. This allows
the creation of a configuration prior to installation of the real system and so can be used to speed
up the installation process.
The configuration created must match the physical equipment in the system into which the
configuration will be loaded. Doing otherwise may cause the system to reset and experience
other problems.
The Create Configuration tool includes all control units, external expansion modules and trunk
cards supported. It is your responsibility to confirm what equipment is supported in your locale.
Creating a New Configuration
Procedure
1. Close or save any configuration currently open.

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 Creating an Offline Configuration

2. Click in the main toolbar or select File | Offline | Create New Config.
3. You should set the Configuration, Locale, Extension Number Length and System Unit
first.
Changing any of these after you start selecting other system hardware will reset the
hardware selections.
4. Select the type of Configuration you want to create.
The other options available will change depending on the selection. If the menu has been
started from Manager running in Server Edition mode, the only option is Server Edition
Edition.
5. Select the Locale for the system.
This defines a range of features such as default telephony settings.
6. The Extension Number Length setting value can be None or 3 to 15.
If a value is selected, all default extension, user and hunt group extension numbers
created by Manager will be that length. In addition Manager will display a warning if an
extension number of a different length is entered when editing the configuration.
7. Select the type of System Unit.
Select the hardware components for the system. For a Server Edition system this is only
necessary if a Expansion System (V2) is selected as the System Units option.
8. Select the additional cards to include in the control unit.
The number and type of cards selectable will depend on the control unit type.
9. Select the external expansion modules to also include in the system.
10. Click OK.
11. For non-Server Edition systems, the configuration is created and loaded into Manager.
For Server Edition systems, the Initial Configuration menu for the selected type of system
unit is displayed. Complete the menu and click Save.
12. Once this configuration has been edited as required it can be saved on the PC or sent to a
system.
13. To Save a Configuration File on the PC Use File | Save Configuration.
14. To Send the Configuration to a System If the system which you want to use the
configuration is available, use File | Offline | Send Configuration to send the configuration
to it.

Warning:
This action will cause the system to reboot and will disconnect all current calls and
service.
• Ensure that you have a copy of the systems existing configuration before overwriting it
with the off-line configuration.

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Editing Configuration Settings

• After sending the configuration, you should receive the configuration back from the
system and note any new validation errors shown by Manager. For example, if using
Embedded Voicemail, some sets of prompt languages may need to be updated to match
the new configurations locale setting using the Add/Display VM Locales option.
Related links
Editing Configuration Settings on page 169

Copying and Pasting

Manager supports the normal Windows methods of cutting, copying, pasting and deleting records
and settings. These can be accessed through the Edit menu in the menu bar or using the
standard Windows keyboard shortcuts for those actions. They can also be accessed by selecting
a record or text field and then right-clicking.
Copy and paste can be used with the navigation and group panes to create a new record with the
same settings as the original. The copy will be renamed as Copy of ... to avoid conflicting with the
original.
When using copy and paste between individual settings fields, whether on the same record or a
different record, care should be taken to ensure that the fields use the same type of data. Similarly
copying a record in the navigation or group pane and then pasting it into the details pane will
prompt Manager to paste the copied records data into the first field of the current record in the
details pane. As a general rule, cut and paste actions should be used with the same pane and
within similar record types.
For users and user rights, a number of controls have been provided to copy settings between a
user and a user right or vice versa. See Configuring User Rights on page 740.
Related links
Editing Configuration Settings on page 169

Saving a Configuration onto PC

The system configuration settings shown within Manager can be saved as a .cfg file on the
Manager PC. These files can be used as backups or sent to other persons to aid problem
diagnostics. Note however that an offline configuration file does not include the Audit Trail records
for the system.
Automatically Saving Sent Configurations
Manager can automatically save a local copies of the configuration. This is done by enabling
either of the Save Configuration File After Load or Backup Files on Send settings (see
Security on page 90).

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 Sending an Individual Configuration

Saving a Configuration Received from a System

1. Select File | Save Configuration as from the menu bar.
2. Saving a Configuration opened on the PC
3. Click in the main toolbar or select File | Save Configuration from the menu bar.
Related links
Editing Configuration Settings on page 169

Sending an Individual Configuration

The current configuration settings open within Manager can be sent to the system. The method
depends on whether Manager is being used to edit the configuration of a single system or a
network of systems.
The first steps of this process depend on whether you are sending a configuration received from
the system or sending one opened offline/created new.

• A Configuration Opened from a System - Click in the main toolbar or select File | Save
Configuration from the menu bar.
• A Configuration Created Offline or Opened from a PC File - Select File | Offline | Send
Config from the menu bar.
The Send Configuration menu is displayed.
Configuration Reboot Mode If Manager thinks the changes made to the configuration settings
are mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge - Send the configuration settings without rebooting the system. This mode should only
be used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate - Send the configuration and then reboot the system.
• When Free - Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Call Barring options.
• Timed - The same as When Free but waits for a specific time after which it then wait for
there to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
• Reboot Time - This setting is used when the reboot mode Timed is selected. It sets the
time for the system reboot. If the time is after midnight, the system's normal daily backup is
canceled.
• Call Barring - These settings can be used when the reboot mode When Free or Timed is
selected. They bar the sending or receiving of any new calls.
Click OK. A service user name and password may be requested.
• If the service user name or password used do not valid, "Access Denied" is displayed.

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Editing Configuration Settings

• If the service user name used does not have rights to send a configuration or to request a
reboot or merge, "Insufficient service user rights" is displayed.
• If the service user name used does not have operator rights to make the changes that have
been made to the configuration, "Insufficient operator rights. Operator cannot modify
xxxx records" is displayed.

• The warning will appear if the configuration being sent contain any errors indicated by a
icon in the error pane. The configuration can still be sent by selected Yes.
• The message Failed to save the configuration data. (Internal error) may indicate that the
IP500 V2 system has booted using software other than that in its System SD card's primary
folder.
Related links
Editing Configuration Settings on page 169

Sending Multiple Configurations

When Manager is running in Server Edition mode or SCN Management mode, it loads multiple
configurations at the same time.

1. Click in the main toolbar or select File | Save Configuration from the menu bar.
2. The menu displayed only shows details for those systems where the system configuration
has been changed and needs to be sent back to the system.

Setting Description
Select By default all systems with configuration changes are selected. If you want to exclude
a system from having its configuration updated, either deselect it or cancel the whole
process.
Change Mode If Manager thinks the changes made to the configuration settings are mergeable, it will
select Merge by default, otherwise it will select Immediate.
Merge Send the configuration settings without rebooting the system. This mode should only be
used with settings that are mergeable. Refer to Mergeable Settings.
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 Erasing the Configuration

Setting Description
Immediate Send the configuration and then reboot the system.
When Free Send the configuration and reboot the system when there are no calls in progress. This
mode can be combined with the Incoming Call Barring and Outgoing Call Barring
options.
Store Offline It is possible to add and edit a configuration file for a system that is not physically
present. Store Offline saves that configuration on the Server Edition Primary in its file
store. The same file is retrieved from there until the physical server is present, at which
time you are prompted whether to use the stored file or the server's current configuration.
Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Incoming Call Barring and Outgoing Call Barring options.
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for the
system reboot. If the time is after midnight, the system's normal daily backup is canceled.
Incoming Call This setting can be used when the reboot mode When Free or Timed is selected. It bars
Barring the receiving of any new calls.
Outgoing Call This setting can be used when the reboot mode When Free or Timed is selected. It bars
Barring the making of any new calls.

3. Click OK. The progress of the sending of each configuration is displayed.

Related links
Editing Configuration Settings on page 169

Erasing the Configuration

About this task
The system configuration settings can be erased. During this process, the system is rebooted
and starts with a set of default settings. This process does not erase the security settings of the
system.
Procedure
1. Select File | Advanced | Erase Configuration (Default).
2. Enter a valid user name and password.
3. The system will be rebooted.
Related links
Editing Configuration Settings on page 169

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Editing Configuration Settings

Default Settings
The following applies to new systems and those defaulted using the Erase Configuration
command. They also apply to IP500 V2 control units defaulted using the reset button on the
rear of the unit (refer to the Installation manual for details of using the reset button).
Mode
IP500 V2 control units can operate in a number of modes. The initial mode is determined by the
type of System SD card fitted and the level of software.
• IP Office A-Law - A system fitted with this type of card will default to A-Law telephony.
• IP Office U-Law - A system fitted with this type of card will default to U-LAW telephony.
• Enterprise Branch - Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch Mode. There is a separate SD card for Enterprise
Branch. The Enterprise Branch SD card can only be used for Enterprise Branch operation
and cannot be used to change modes to IP Office. You also cannot use or change an IP
Office SD card for use with an Enterprise Branch system.
- Do not re-purpose a Enterprise Branch card for use with any other IP Office mode. Doing
so may damage the SD card and make it unusable for your Enterprise Branch system.
Default Short Codes
For IP500 V2 control units, A-Law or U-Law operation is determined by the Feature Key dongle
installed in the system. Depending on the variant, a default system will use different sets of default
short codes. See the Default System Short Code List on page 947.
Default Data Settings

When a new or defaulted control unit is switched on, it requests IP address information from a
DHCP Server on the network. This operation will occur whether the LAN cable is plugged in or
not.
If a DHCP server responds within approximately 10 seconds, the control unit defaults to being a
DHCP client and uses the IP address information supplied by the DHCP server.
If no DHCP Server responds, the control unit still defaults to being the DHCP client but assumes
the following default LAN addresses:
• For its LAN1 it allocates the IP address 192.168.42.1 and IP Mask 255.255.255.0.
• For its LAN2 if supported, it allocates the IP address 192.168.43.1 and IP Mask
255.255.255.0.
Once a control unit has obtained IP address and DHCP mode settings, it will retain those settings
even if rebooted without a configuration file present on the System SD card. To fully remove the
existing IP address and DHCP mode setting, the system must be defaulted using Manager.
Default Security Settings
Security settings are held separately from the configuration settings and so are not defaulted by
actions that default the configuration. To return the security settings to their default values the
separate Erase Security Settings command should be used.

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 Default Settings

Default Standard Mode Telephony Configuration Settings

• A hunt group Main is created with extension number 200. The first 16 extensions on the
systems are added to the group.
• All physical extensions ports are numbered from extension number 201 upwards. A matching
user record for each extension is also created.
• A default incoming call route for all voice calls is created, with the default hunt group Main as
its destination.
• A default incoming call route for data calls is created with the default RAS record DialIn as its
destination.
• All lines are defaulted to Incoming Group ID and Outgoing Group ID of 0.
• Default short codes are created based on whether the system's locale is A-Law or U-Law.
• Default Server Edition Telephony Configuration Settings
• No users except NoUser.
• All extensions are unnumbered.
• No default hunt group or incoming calls routes are created.
• All auto-create options are off by default.
Related links
Editing Configuration Settings on page 169

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Chapter 19: Creating New Records

There are a number of ways in which you can add new records to the configuration currently loaded
in Manager.

Adding a New Record Using the Details Pane

Procedure
1. Use the navigation pane, groups pane or navigation toolbar, select an existing record of
the type required.
2. Click at the top-right of the details pane.
3. Select the type of record required. For example, with extensions you can select from H.323
Extension or SIP Extension.
4. Complete the settings for the new record and click OK.

Adding a New Record Using the Group Pane

Procedure
1. Using the navigation pane or navigation toolbar, select the type of record required.

2. Right-click on the group pane and select New.

3. If a list is displayed, select the specific type of record required.
4. Complete the settings for the new record and click OK.

Adding a New Record Using the Navigation Pane

Procedure
1. In the navigation pane, right-click on the type of record required and select New.
2. If a list is displayed, select the specific type of record required.

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 Other record creation methods

3. Complete the settings for the new record and click OK.

Other record creation methods

Procedure
1. Import records
• New records can be created for users, hunt groups, short codes and directory entries by
importing files. See Importing and Exporting Settings on page 190.
2. Automatically create records
There are scenarios where new records are automatically added to the configuration:
• Certain default records are added to the configuration when a new system is started or
when a system configuration is erased.
• New records for extensions and lines are automatically added to match the hardware
present when a system is started or rebooted.
• For telephones connected via IP links (H.323, SIP and DECT), the system can
be configured to automatically create extension and or user records to match the
telephone.
3. Use templates to create records
• Manager supports a number of methods by which new records can be created from
templates. See Working with Templates on page 686.

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Chapter 20: Importing and Exporting
Settings

Manager can import configuration settings created elsewhere. This can be useful when setting up a
new system or sharing common settings such as a directory between systems.
Settings are imported and exported in the following formats:
File Type Description
Binary Files (.exp) These are non-editable files. During export, it is possible to select what types of records
are included in the file. During import, the whole file is imported. The types of records
supported are:
• ARS, Control Unit, Extension, Firewall Profile, Group, Incoming Call Route, Line, RAS,
System Short Codes, Users, User Rights, Account Codes, Authorization Codes, Auto-
Attendants, Conferences, Directory, IP Route, License, Location, Logical LAN, RAS
Location Request, Service, Time Profile, Tunnel, WAN Port.
Comma Separated These are editable plain text files. The types of records supported are:
Variable Text Files
• Groups, Users, Directory, System Short Codes, Licenses (ADI only), Full Configuration.
(.csv)
• Only key values for the records selected are included. See the table below for details.

CSV File Format

The format is CSV using commas as field separator, no text delimiters and no header row. The
simplest way to check the required format for a CSV file before import, is to export a file from an
existing system configuration.
• UTF-8 Character Encoding
Manager imports and exports CSV files using UTF-8 character encoding which uses a multiple
bytes to support characters with diacritic marks such as ä. Other applications, such as Excel,
may, depending on the user PC settings, use different encoding which will cause such
characters to be removed or corrupted. Care should be taken to ensure that any tool used
to create or edit the CSV supports all the characters expected and uses UTF-8 format.
File Name Fields in Order
Configuration The format of this CSV is too complex to be described. This file format
should only be used for export and import between systems and not for any
offline editing.
License This option is only available for export and only for ADI licenses present on
the system.
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 Exporting Settings

File Name Fields in Order

Short Code Code, Telephone Number, Feature.
User Name, Extension, User Rights, Email Address, Full Name, Password,
VoiceMail Code, Login Code, User Template, Extension Template.
Directory Name, Number, Speed Dial.
Group Name, Extension, Group, Hunt, Rotary, Longest Waiting, Queuing On,
Voicemail On, Broadcast, Voicemail Email.
• Apart from the Name, Extension and Voicemail Email fields, all other
fields use a 1 or 0 value for on or off.

Excel Notes
When opening a .csv file in Excel, it will alter the way some data is displayed, for example
automatically changing the display format of dates and long numbers such as phone numbers.
Therefore it is important to use the following steps when using Excel.
• Importing into Manager from Excel From Excel, save the file as a .csv. This file will use ANSI
character encoding. Open the file in Notepad and use the Save As option to rename the file
and select UTF-8 encoding. Import the UTF-8 version of the file into Manager.
• Exporting from Manager into Excel Do not double-click on the file exported from Manager.
Start Excel and use File | Open to select the file. Excel will recognize that the file uses UTF-8
encoding and will start its text file importation wizard. Follow the wizard instructions and select
comma as the field delimiter.
Using the CSV Configurator spreadsheet
You can use the CSV Configurator spreadsheet to create or modify multiple configuration entries.
The CSV Configurator spreadsheet IP Office User CSV Configurator.xlsm is available in
the IP Office Manager application folder.

Exporting Settings
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Export.
3. Select the type of file. The list of exportable record types will change to match the file type.
4. Select the types of items that should be exported.
5. Use the Save In path to select the location for the exported files.
• The default location used is a sub-folder in the Manager application directory based
on system name of the currently loaded system. For example, ...\Avaya\IP
Office\Manager\System_1.
6. Click OK.

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Importing and Exporting Settings

Importing Settings
Importing settings will overwrite any existing records that match a record being imported
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Import.
3. Select the type of file. The list of items will change to match the type of file selected and
whether a matching file or files is found in the current file path.
4. Use Look In to adjust the file path.
• The default location used is a sub-folder in the Manager application directory based
on system name of the currently loaded system. For example, ...\Avaya\IP
Office\Manager\System_1.
5. Select the types of items that should be imported.
6. Click OK.

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Chapter 21: Known System Discovery

The Select IP Office menu normally displays systems discovered by Manager using either UDP
broadcast and or TCP requests. Manager can be configured to also record details of discovered
units and then display a list of those previously discovered ('known') systems.

Configuring Manager for Known System Discovery

About this task
Use of known systems discovery is not enabled by default. The Manager must be configured for
the feature with a file location to which it can store and retrieve known system details.
Procedure
1. Select File | Change Working Directory.

2. In the Known Units File field, enter the directory path and file name for a CSV file into
which Manager can write details of the systems it discovers.
If the file specified does not exist it will be created by Manager.
3. Click OK.

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Known System Discovery

Using Known System Discovery

Procedure
1. When the Select IP Office screen is displayed click on Known Units.

2. The screen displays the list of systems previously discovered and stored in the CSV file.
3. To select an control unit, highlight the row containing unit data and click OK.
The selected unit will appear in the Select IP Office window.
4. To filter displayed units, type the first few characters of the unit name in the Filter field.
Any unit whose name does not match the filter will be temporarily hidden.
5. Each discovery appends data to the known unit list.
It is possible that details of some records in the list may be out of date. Right clicking on
the leftmost (grey) column of any row will bring up a floating menu offering the options of
Refresh and Delete.
6. A new record may be manually added without having to access the system first through
normal discovery.
Enter the IP address of the new system in the IP Address column of the blank row shown
with a * and select Refresh from the floating menu. This will update the Known Units file
with data relating to the unit with the specified address.
7. Select Cancel to return to the Select IP Office menu.
Result
Note:
• The key used by the Known Systems CSV file is the IP address. The file cannot contain
records for separate systems that use the same IP address for access.
• The file can be made read only. In that case any attempts using Manager to update the
file will be ignored.

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Part 5: System Configuration
Settings

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Chapter 22: Configuration Mode Field
Descriptions

This following sections detail the configuration settings for the different record types within the
system. Depending on the type and locale of the system some settings and tabs may be hidden as
they are not applicable. Other settings may be grayed out. This indicates that the setting is either for
information only or that another setting needs to be enabled first.
Related links
Configuration field display in Standard mode on page 196
Configuration field display in Server Edition mode on page 197

Configuration field display in Standard mode

This order of display of different types of configuration record is used for Standard mode systems.
Icon Description
System - Overall settings for the data and telephony operation of the system.

Line - Settings for trunks and trunk channels within the system.

Control Unit - Information summary of the system.

Extension - Settings for extension ports.

User - Settings for each system user. They may or may not be associated with an extension.

Hunt Group - Collections of users to which calls can be directed for answer by any one of those
users.
Short Code - These are numbers which when dialed trigger specific features or are translated for
external dialing. Short codes can be set at both the system wide level and locally for a particular
system.
Service - Configuration settings such as user names and passwords needed for connections to data
services such as the Internet.
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 Configuration field display in Server Edition mode

Icon Description
RAS - Remote Access Service settings for connecting incoming data calls.

Incoming Call Route - Records here are used to match incoming call details on external trunks to
destinations on the system.
WAN Port - Configuration settings for the WAN ports provided on some units.

Directory - External names and numbers. Used for matching names to incoming calls and for
dialing from user applications.
Time Profile - Used to control when various functions are active.

Firewall Profile - Use to control the types of data traffic that can cross into or out of the system.

IP Route - These records are used to determine where data traffic on the system should be routed.

Account Code - Used for call logging and to control the dialing of certain numbers.

License - License keys are used to enable system features and applications.

Tunnel - Used to created IPSec and L2TP data tunnels.

User Rights - Provide templates to control the settings applied to associated users.

Auto Attendant - Used when an Avaya memory card is installed in the control unit.

ARS - Automatic Route Selection is used by to control outgoing external calls.

Authorization Codes - Authorization codes are similar to account codes. However, unlike account
codes which are usable by any user, each authorization code is only usable by a specific user or
users associated with a specific set of user rights.

Related links
Configuration Mode Field Descriptions on page 196

Configuration field display in Server Edition mode

When Manager is being used in Server Edition mode, the navigation pane works in the same
way as normal. However, the different types of configuration records are ordered and grouped
differently. This reflects the fact that some types of record are automatically shared across all
systems in the network.
Solution settings
The first 8 types are records have special behaviors that are different from normal records stored
in the configurations of individual systems in the network.

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Configuration Mode Field Descriptions

Icon Description
Directory - External names and numbers. These records are used to match names to incoming
calls and for making calls by name selection from the directory on phones or in applications. These
directory records are stored in the configuration of the Primary Server. By default all other systems in
the network automatically import a copy of the Primary Server system directory at regular intervals.
Hunt Group - These records are groups of users to which calls can be directed for answering by
any one of those users. Hunt group records are stored in the configuration of the Primary Server but
those hunt groups are advertised for use by all systems in the network.
User - These records show settings for system users. Each user may or may not be associated with
an extension. All the users configured on all systems are grouped here to allow easy configuration
access. The individual user records are still stored in the configuration of the particular system on
which the user was created and can also be accessed through that system's configuration settings.
New users are created through the User settings of the system that hosts the user.

By default, the following types of records are shared and replicated by each system in the
network and cannot be set at an individual system level. That operation can be changed using
the consolidation settings.
Icon Description
Short Code - These are numbers which when dialed trigger specific features or are translated for
external dialing. These short codes are common to all systems in the network.
Incoming Call Route Records - These set here are used to match incoming call details on external
trunks to destinations. These incoming call routes are shared by all systems in the network.
Time Profile - Used to control when various functions are active. The time profiles set here are
shared by all systems in the network.
Account Code - Used for call logging and to control the dialing of certain numbers. The account
codes set here are shared by all systems in the network.
User Rights - Provide templates to control the settings applied to users associated with a particular
set of user rights. These user rights are shared and replicated on all systems in the network.

Individual system settings

In addition to the settings above, a range of other types of record can be configured for each
individual system in the network. Visibility and configuration of Short Code, Incoming Call Route,
Time Profile, Account Code and User Rights records is dependent on the consolidation settings
of Manager.
Icon Description
System - A system icon is shown for each system in the network. That is, one for the Primary
Server, one for the the Secondary Server if installed and one for each Expansion System (L) and
Expansion System (V2) systems. Each can be expanded to allow configuration of records that are
particular to that system.
Line - Settings for trunks and trunk channels within the system.

Control Unit - Information summary of the system.

Table continues…

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 Configuration field display in Server Edition mode

Icon Description
Extension - Settings for extension ports.

User - Settings for each system user. They may or may not be associated with an extension.

Short Code - These are numbers which when dialed trigger specific features or are translated for
external dialing.
Service - Configuration settings such as user names and passwords needed for connections to data
services such as the Internet.
RAS - Remote Access Service settings for connecting incoming data calls.

WAN - Port Configuration settings for the WAN ports provided on some units.

Firewall Profile - Use to control the types of data traffic that can cross into or out of the system.

IP Route - These records are used to determine where data traffic on the system should be routed.

License - License keys are used to enable system features and applications.

ARS - Automatic Route Selection is used by to control outgoing external calls.

Authorization Codes - Authorization codes are similar to account codes. However, unlike account
codes which are usable by any user, each authorization code is only usable by a specific user or
users associated with a specific set of user rights.

Related links
Configuration Mode Field Descriptions on page 196

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Chapter 23: BOOTP Record

Navigation: BOOTP | BOOTP Record

The BOOTP settings are used by the Manager application itself. They are not system configuration
settings.
BOOTP is protocol used by devices to request software when restarting. It is used when upgrading
the control unit within a system or when the core software within the control unit has been erased.
When running, Manager can respond to BOOTP requests and, if it finds a matching BOOTP record
for the system, provide the software file indicated by that record.
BOOTP records are not part of a system's configuration settings, they are items saved on the
Manager PC. Normally Manager automatically creates a BOOTP record for each system with which
it has communicated, up to a maximum of 50 records. However BOOTP records can be added and
edited manually when necessary.
• The location from which Manager provides files in response to BOOTP is its binaries directory.
This can be changed using File > Change Working Directory or File > Preferences >
Directories. This directory is also the directory used by Manager when providing files by TFTP.
• Manager can be disabled from providing BOOTP support for any systems. Select File >
Preferences > Preferences > Enable BOOTP and TFTP Server.
Field Description
Enabled Default = Enabled
If unchecked, BOOTP | BOOTP Record support for the matching system from this
Manager PC is disabled.
System Name This field is not changeable. It shows the system name.
MAC Address The MAC address of the system. The address can be obtained and or verified in a number
of ways:
• When a system's configuration settings are loaded into Manager, it is shown as the
Serial Number on the Unit form. On defaulted systems, it is also used as the system
name.
• If the system is requesting software, the MAC address is shown as part of the request in
the status bar at the base of the Manager screen.
• If the system can be pinged, it may be possible to obtain its MAC address using the
command arp -a <ip address>.
IP Address The IP address of the system's LAN1.
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Field Description
Filename The name of the .bin software file used by that type of control unit. To be transferred to the
system, this file must exist in the Manager applications Working Directory.
Time Offset Default = 0.
In addition to performing BOOTP support for systems, the Manager application can also
act as a time server (RFC868). This field sets the offset between the time on the PC
running Manager and the time sent to the system in response to its time requests. The
field is not used if a specific Time Server IP Address is set through the System form in the
system's configuration settings.
Manager can be disabled from acting as an Internet Time (RFC868) server. Select File >
Preferences > Preferences and uncheck Enable time server.

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Chapter 24: Operator

Operator records are not part of a system's configuration settings. They are used when a pre-
Release 3.2 configuration is loaded to control what parts of a configuration can be edited.
Operator View Edit New Delete Configuration Record Types
Administrator ✓ ✓ ✓ ✓ All configuration records
Manager ✓ ✓ ✓ ✓ View all. Other actions
Extension, User, Hunt Group,
Short Code, Service, RAS,
Incoming Call Route, Directory,
Time Profile, Firewall Profile,
IP Route, Least Cost Routing,
Account Code, ARS.
Operator ✓ ✓ – – View all configuration records.
Edit all except System, Line,
Control Unit and Authorization
Codes.
Guest ✓ – – – View all.

If an invalid operator is specified while receiving a configuration from a pre-3.2 system, the settings
will be loaded using the Guest operator.

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Chapter 25: System

Navigation: System
There is one System record for each system being managed. When managing multi system Server
Edition or Small Community Network deployments, clicking on the System icon for a particular
system displays a system inventory page for that system.
Related links
System on page 203
LAN1 on page 213
LAN2 on page 230
DNS on page 230
Voicemail on page 232
Telephony on page 239
Directory Services on page 259
System Events on page 265
SMTP on page 271
SMDR on page 272
VCM on page 274
Integrated Contact Reporter on page 276
VoIP on page 277
Dialer on page 282
Contact Center on page 284
Remote Operations on page 286
Avaya Cloud Services on page 286
Avaya Push Notification Services on page 289

System
Navigation: System | System
Additional configuration information
For additional information on time settings, see System Date and Time on page 663.

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System

Configuration settings
These settings are mergeable except Locale and Favor RIP Routes over Static Routes.
Changing those settings requires a reboot of the system.
Field Description
Name Default: = System MAC Address.
A name to identify this system. This is typically used to identify the configuration by the
location or customer's company name. Some features such as H.323 Gatekeeper require
the system to have a name.
• This field is case sensitive and within any network of systems must be unique.
• Do not use <, >, |, \0, :, *, ?, . or /.
Contact Default = Blank.
Information
This field is only be edited by service user with administrator rights. If a value is entered, it
sets the system under 'special control'.
If the contact information is set using a standalone version of Manager, warnings
that "This configuration is under special control" are given when the
configuration is opened again. This can be used to warn other users of Manager that the
system is being monitored for some specific reason and provide them with contact details
of the person doing that monitoring.
Locale Sets default telephony and language settings based on the selection. It also sets various
external line settings and so must be set correctly to ensure correct operation of the
system. See Avaya IP Office Locale Settings.
• For individual users, the system settings can be overridden through their own locale
setting Select User | User | Local.
Location Default = None.
Specify a Location entry for the system. This location is then used as the default
Location settings for all the system's extensions and lines unless they are specifically
configured with a different location. See Using Locations on page 617.
• If Location entries have been defined, a location must be assigned to the system and
to all systems in the network.
Customize Locale Settings
The Customize locale matches the Saudi Arabia locale but with the following additional controls shown below.
For other locales, these are set on System | Telephony | Tones and Music.
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 System

Field Description
Tone Plan Default = Tone Plan 1
The tone plan control tones and ringing patterns. The options are:
• Tone Plan 1: United States.
• Tone Plan 2: United Kingdom.
• Tone Plan 3: France.
• Tone Plan 4: Germany.
• Tone Plan 5: Spain.
CLI Type Used to set the CLI detection used for incoming analog trunks. The options are: DTMF,
FSK BELL202, and FSK V23.

Device ID Server Edition only. Displays the value set for Device ID on the System > System
Events > Configuration tab.
If SSL VPN is configured, Avaya recommends that the Device ID matches an SSL VPN
service Account Name. Each SSL VPN service account name has an associated SSL
VPN tunnel IP address. Having the displayed Device ID match an SSL VPN service
account name helps identify a particular SSL VPN tunnel IP address to use for remotely
managing the IP Office.
TFTP Server IP Default = 0.0.0.0 (Disabled). On Server Edition Systems, the default on Secondary and
Address Expansion servers is the Primary Server address.)
If the Phone File Server Type below is set to Custom, this address is included as the
TFTP file server address sent in the system’s DHCP response to phones.
• You can use the address 255.255.255.255 to broadcast for the first available TFTP
server on the network.
• IP Office Manager can act as a TFTP server to provide files from its configured
binaries directory. This requires the IP Office Manager setting File > Preferences >
Preferences > Enable BootP and TFTP Servers enabled.
• On IP500 V2 systems, you can enter the LAN1 IP Address to use the system’s own
memory card as the TFTP file source. This requires the security setting Unsecured
Interfaces > Applications Controls > TFTP Directory Read enabled.
HTTP Server IP Default = 0.0.0.0 (Disabled).
Address
This address, if set, is used in a number of scenarios:
• DHCP Responses: If the Phone File Server Type below is set to Custom, this
address is included as the HTTP file server address sent in the system’s DHCP
response to phones.
• HTTP Redirection: If HTTP Redirection below is enabled, 9608, 9611, 9621, 9641,
and H.323 phone binary file requests sent to the system are redirected to this address.
• B199/H175 Phones/Vantage Phones: Phone firmware file requests sent to the system
from these types of phone are always redirected to this address (B199 phones running
R1.0 FP6 or higher).
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Field Description
HTTP Server URI Default = Value provided by the deployment’s Customer Operations Manager.
Used by subscription mode systems.
• If set, software file requests from Avaya Workplace Client and Vantage phones are
redirected to this address.
• If not set, then the Avaya Workplace Client and Vantage phones use the HTTP Server
IP Address setting.
Phone File Server Default = Memory Card (IP500 V2)/Disk (Linux system).
Type
For IP phones (H.323 and SIP) using the system as their DHCP server, the DHCP
response can include the address of a file server from which the phone should request
files. The setting of this field controls which address is used in the DHCP response. The
options are:
• Custom
The DHCP response the system provides to phones contains the addresses set in the
TFTP Server IP Address and HTTP Server IP Address fields.
• Disk: (Linux systems only)
The system uses its hard disk for file requests from phones. The DHCP response the
system provides to phones contains its the LAN address as the TFTP and HTTP file
server address.
• Memory Card: (IP500 V2 only)
The system uses it memory card for file requests from phones.The DHCP response the
system provides to phones contains its the LAN address as the TFTP and HTTP file
server address. This is supported for up to 50 IP phones total.
• Manager: (IP500 V2 only)
The system forwards phone file requests to the configured Manager PC IP Address
set below. The DHCP response the system provides to phones contains the system’s
LAN address as the HTTP file server address.
- HTTP-TFTP Relay is support when using IP Office Manager as the TFTP server
(not supported by Linux based systems). This is done by setting the TFTP Server
IP Address to the address of the IP Office Manager PC and the HTTP Server IP
Address to the control unit IP address. This method is supported for up to 5 IP
phones total.
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 System

Field Description
HTTP Redirection Default = Off.
For some phones using the IP Office as the file server, their request for firmware files can
be redirected to another file server. This is useful when the firmware files are large or to
enable multiple IP Office systems to share a common firmware file server.
When enabled, firmware file requests are redirected to the address set by the HTTP
Server IP Address field. That field is available when the Phone File Server Type is set
to Memory Card or Disk.
IP Office HTTP redirection is only supported for the following phones:
• 9600 Series and J100 Series phones.
• B199, H175 and Vantage phone firmware requests are always redirected to the HTTP
Server IP Address regardless of the HTTP Redirection and Phone File Server Type
settings.
- For R11.1.2.4, this is also applied to B199 phones running R1.0 FP6 or higher
firmware.
Manager PC IP Default = 0.0.0.0 (Broadcast).
Address
This address is used when the Phone File Server Type is set to Manager.
Avaya HTTP Default = Off.
Clients Only
When selected, the IP Office only responds to HTTP requests from another IP Office
system, Avaya phone, or Avaya application.
Enable SoftPhone Default = Off.
HTTP Provisioning
This option must be enabled if the IP Office Video Softphone is being supported.
Use Preferred Default = Off
Phone Ports
Set the ports indicated in the auto-generated 46xxsettings.txt file requested by
phones.
• When not enabled:
IP Office addresses in the auto-generated 46xxsettings.txt file use ports 80
(HTTP) and 443 (HTTPS).
• When enabled:
IP Office addresses in the auto-generated 46xxsettings.txt file uses ports 8411
(HTTP) and 411 (HTTPS).
Regardless of the setting, the IP Office will accept requests on HTTP 80 and HTTPS 443.
This is required for legacy phones that do not use the 46xxsettings.txt file settings
and to redirect existing phones to the preferred phone ports.
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Field Description
Favor RIP Routes Default = Off
over Static Routes
You can enabled RIP on the LAN1 and LAN2 interfaces and on specific Services. This
setting controls how the IP Office system uses a RIP route when it has a static route to
the same destinations configured in the IP Routes settings. This option is not supported
on Linux-based systems.
• When enabled:
RIP routes to a destination override any static route to the same destination. This
applies even if the RIP route has a higher metric.
- The exception is RIP routes with a metric of 16 which are always ignored.
- If a learned RIP route fails, the IP Office applies a metric of 16 for five minutes after
the failure.
• When disabled:
RIP routes to destinations which have static routes configured are ignored.
Automatic Backup Default = On.
This command is available with IP500 V2 systems. When selected, as part of its daily
backup process, the system automatically copies the folders and files from the System
SD card's /primary folder to its /backup folder. Any matching files and folders already
present in the /backup folder are overwritten.
• On subscription mode systems, COM supports a separate daily backup of configuration
settings.
Media Archival For subscription mode systems, this field sets with application is used as the voice
Solution recording library (VRL) application for call recordings:
• Local Media Manager
Use the media manager service running locally on the same server as the voicemail
service. Refer to the Administering Avaya IP Office™ Platform Media Manager.
• Centralized Media Manager
Use the media manager service provided by the same cloud based services providing
the system subscriptions.
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Field Description
Messaging server This field sets which service is used as the instant messaging server for Avaya
applications. The following options are supported:
• one-X Portal
Use the system's Avaya one-X® Portal for IP Office server for instant messaging
between IP Office clients, including Avaya Workplace Client.
- This method is not supported for Avaya Workplace Client users logging in using SSO
or email. User's must register directly to the IP Office system.
• Avaya Spaces
Use Avaya Spaces for instant messaging for Avaya Workplace Client users. It does not
include non-Avaya Spaces users.
- This requires the Avaya to be configure to support Avaya Cloud Services. For details,
see the IP Office Avaya Workplace Client Installation Notes manual.
- This method does not support sending push notifications for instant messages. That
is, instant messages are not received by iOS clients when the client is suspended or
in the background.
- Not supported for remote Android/iOS Avaya Workplace Client using IPv6.
Provider Default = Not visible.
This field is visible if the system has been branded by addition of a special license for a
specific equipment provider.
• The branding is fixed, that is it remains even if the license is subsequently removed.
• The number shown is a unique reference to the particular equipment provider for whom
the system has been branded.
• When branded, the equipment provider's name is displayed on idle phone displays and
other provider related features are enabled.
Reseller This field is shown on subscription mode systems. The value is automatically set when
the system is first subscribes.

Warning:
• Do not change the value except under guidance from Avaya. Changing the value
can cause lose of the system's subscriptions and remote management services
through COM.
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Field Description
Time Setting Time and date settings are only shown for IP500 V2 based systems. The time and date
Config Source for Linux-based servers are set through the server’s Platform View menus.

Important:
An accurate time source and settings are vital to many functions, including any
services that use certificates. Avaya recommend that you use SNTP and a reliable
source such as time.google.com.
• SNTP
Use a list of SNTP servers to obtain the UTC time. The IP Office tries the listed servers
in order until it receives a response. The IP Office makes a request following a reboot
and every hour afterwards.
- In a network, other IP Office servers can use the primary IP Office as their SNTP
server.
• Voicemail Pro/Manager (Obsolete)
Both Windows-based Voicemail Pro and IP Office Manager can act as RFC868 Time
servers for the IP Office. Use of other RFC868 server sources is not supported.
They provide both the UTC time value and the local time as set on the PC. The
system makes a request to the specified address following a reboot and every 8 hours
afterwards.
• None
Enable users with System Phone Rights (User > User) to set the time and date
from their own extension. The IP Office can still apply daylight saving settings to the
manually set time.
File Writer IP Default = 0.0.0.0 (Disabled)
Address
This field set the address of the PC allowed to send files to the System SD card installed
in the system using HTTP or TFTP methods other than embedded file management.
• On non-Linux based systems, this field sets the address of the PC allowed to send
files to the memory card using HTTP or TFTP methods other than embedded file
management.
• For Linux based systems it is applied to non-embedded file management access to
the /opt/ipoffice folder on the server.
An address of 255.255.255.255 allows access from any address. If embedded file
management is used, this address is overwritten by the address of the PC using
embedded file management (unless set to 255.255.255.255).
Dongle Serial Displayed ony for pre-Release 10.0 IP500 V2 systems using ADI licensing. For system’s
Number using PLDS licensing, see the PLDS Host ID (License > License).
This field is for information only. It shows the serial number of the feature key dongle
against which the system last validated its licenses. Local is shown for a serial port,
Smart Card or System SD feature key plugged directly into the control unit. Remote is
shown for a parallel or USB feature key connected to a feature Key Server PC. The serial
number is printed on the System SD card and prefixed with FK.
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Field Description
System Displayed for Linux based systems. This field is for information only.
Identification
This is the unique system reference that is used to validate licenses issued for this
particular system. For a physical server this is a unique value based on the server
hardware. For a virtual server this value is based on several factors including the LAN1
and LAN2 IP addresses, the host name and the time zone. If any of those are changed,
the System ID changes and any existing licenses become invalid.
AVPP IP Address Default = 0.0.0.0 (Disabled)
Where Avaya 3600 Series SpectraLink wireless handsets are being used with the system,
this field is used to specify the IP address of the Avaya Voice Priority Processor (AVPP)

Time Setting Config Source = None/SNTP

These settings are shown for IP500 V2 based systems where the Time Setting Config Source
has been set to None or SNTP. The time, date and timezone for Linux-based servers are set
through the server Platform View menus.
• If Location entries have been defined, a location must be assigned to the system. The
location's time settings (other than time source) override the settings below unless set to
Same as System.
Field Description
Time Server Default = Blank
Address
Displayed when the Time Setting Config Source is set to SNTP. For the SNTP servers,
enter a list of IP addresses, host names, or fully-qualified domain names (FQDN).
Separate each record with a space. The use of broadcast addresses is not supported.
• The list is used in order of the records until a response is received.
• In a network, other IP Office servers can use the primary IP Office as their SNTP server.
Time Zone Select a time zone from the list. This sets the default time offset and DST to match the
chosen time zone.

Field Description
Local Time Offset Default = Based on the selected locale and time zone. See Avaya IP Office Locale
from UTC Settings.
This setting is used to set the local time difference from the UTC time value provided by
SNTP. For example, if the system is 5 hours behind UTC, configured this field as -05:00.
• You can adjust the offset in 15 minute increments.
Use this offset for the standard (non-daylight savings time) time. To apply an additional
offset for daylight saving time periods, using the settings below.
Automatic DST Default = Based on the selected locale and time zone. See Avaya IP Office Locale
Settings.
When enabled, the system automatically corrects for daylight saving time (DST) changes
using the settings below.
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System

Field Description
Clock Forward/ Default = Based on the selected locale and time zone. See Avaya IP Office Locale
Back Settings Settings.
This field displays entries for when the IP Office should apply and remove a daylight
saving time offset in addition to the Local Time Offset from UTC.
You can configure up to 10 entries (20 for IP Office R11.1.3.2 and higher).
• To edit an entry, select it and then click Edit.
• To delete an entry, select it and click Delete.
• In order to add a new entry you may need to delete an existing entry. The option Add
New Entry then appears at the bottom of the list.
Each entry has the following settings:

Field Description
DST Offset The number of hours to shift the local time for DST.
Clock Select Clock Forward to see and edit when the clock will move
Forward/Back forward to start daylight saving.
Select Clock Back to see and edit when the clock will move
backward to end daylight saving.
Local Time To The time of day to move the clock forward to start daylight saving.
Go Forward

Local Time To The time of day to move the clock backward to end daylight saving.
Go Back
Date for Clock The date for moving the clock forwards or backwards. Select the date
Forward/Back by double-clicking on it in the calendar.

Time Setting Config Source = Voicemail Pro/Manager

These settings are shown for IP500 V2 based systems where the Time Setting Config Source
has been set to Voicemail Pro/Manager.
Field Description
IP Address Default = 0.0.0.0
The address to which the IP Office should send time requests. This must be the address
of an server running Voicemail Pro or IP Office Manager.
• When set to 0.0.0.0, following a reboot the IP Office first makes the request to the
Voicemail Server IP address if set and, if it receives no reply, then makes a broadcast
request.
• For Windows-based Voicemail Pro servers, if IP Office Manager is already running on
the server when the voicemail service starts, voicemail will not act as a time server.
• You can stop IP Office Manager acting as an RFC868 time server by deselecting the
Enable Time Server option (File > Preferences > Preferences).
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 LAN1

Field Description
Time Offset Default = 00:00.
This value is not normally set as the IP Office matches any time changes, including
daylight savings, that occur on the time source PC.

Related links
System on page 203

LAN1
Navigation: System | LAN1
Used to configure the behavior of the services provided by the system's first LAN interface.
Up to 2 LAN's (LAN1 and LAN2) can be configured. The control unit has 2 RJ45 Ethernet ports,
marked as LAN and WAN. These form a full-duplex managed layer-3 switch. Within the system
configuration, the physical LAN port is LAN1, the physical WAN port is LAN2.
Configuring both interfaces with the same IP address on the same subnet is not supported.
However, no warning is issued when this configuration is implemented.
Related links
System on page 203
LAN Settings on page 213
VoIP on page 215
Network Topology on page 223
DHCP Pools on page 228

LAN Settings
Navigation: System | LAN | LAN Settings
Used to set the general LAN settings for the LAN interface such as the IP address mode.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
IP Address Default = 192.168.42.1 or DHCP client.
This is the IP address of the Control Unit on LAN1. If the control unit is also acting as
a DHCP server on the LAN, this address is the starting address for the DHCP address
range.
IP Mask Default = 255.255.255.0 or DHCP client.
This is the IP subnet mask used with the IP address.
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Field Description
Primary Trans. IP Default = 0.0.0.0 (Disabled)
Address
This setting is only available on control units that support a LAN2. Any incoming IP
packets without a service or session are translated to this address if set.
RIP Mode Default = None.
Routing Information Protocol (RIP) is a method by which network routers can exchange
information about device locations and routes. Routes learnt using RIP are known as
'dynamic routes'. The system also supports 'static routes' though its IP Route records.
For Server Edition systems this setting is only available on Expansion System (V2)
systems. The options are:
• None: The LAN does not listen to or send RIP messages
• Listen Only (Passive): Listen to RIP-1 and RIP-2 messages in order to learn RIP
routes on the network.
• RIP1: Listen to RIP-1 and RIP-2 messages and send RIP-1 responses as a sub-
network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): Listen to RIP-1 and RIP-2 messages and
send RIP-2 responses as a sub-network broadcast.
• RIP2 Multicast: Listen to RIP-1 and RIP-2 messages and send RIP-2 responses to
the RIP-2 multicast address.
Enable NAT Default = Off
This setting controls whether NAT should be used for IP traffic from LAN1 to LAN2. This
setting should not be used on the same LAN interface as a connected WAN3 expansion
module.
Number of DHCP IP Default = 200 or DHCP client. Range = 1 to 999.
Addresses
This defines the number of sequential IP addresses available for DHCP clients.
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Field Description
DHCP Mode Default = DHCP Client.
This controls the control unit's DHCP mode for the LAN. When doing DHCP:
• LAN devices are allocated addresses from the bottom of the available address range
upwards.
• Dial In users are allocated addresses from the top of the available range downwards.
• If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users are
allocated their address from the LAN1 pool of addresses first.
The options are:
• Server: When this option is selected, the system will act as a DHCP Server on this
LAN, allocating address to other devices on the network and to PPP Dial in users.
• Disabled When this option is selected, the system will not use DHCP. It will not act as
a DHCP server and it will not request an IP address from a DHCP server on this LAN.
• Dial In When this option is selected, the system will allocate DHCP addresses to PPP
Dial In users only. On systems using DHCP pools, only addresses from a pool on the
same subnet as the system's own LAN address will be used.
• Client When this option is selected, the system will request its IP Address and IP
Mask from a DHCP server on the LAN.

Note:
Do not use this option with a limited time lease line.
• Advanced: The system can be configured with a number of DHCP Pools from which
it can issue IP addresses.

Related links
LAN1 on page 213

VoIP
Navigation: System | LAN | VoIP
Configuration settings
Used to set the system defaults for VoIP operation on the LAN interface.
The following settings are mergeable:

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System

• Auto-create Extn • Scope

• Auto-create User • Initial keepalives
• H.323 Signaling over TLS • Periodic timeout
• Remote Call Signaling Port • VLAN
• Auto-create Extn/User • 1100 Voice VLAN Site Specific Option Number
(SSON)
• Enable RTCP Monitoring on Port 5005
• 1100 Voice VLAN IDs
• RTCP collector IP address for phones

The remaining settings are not mergeable. Changes to these settings requires a reboot of the
system.
H.323 Gatekeeper Enable
Field Description
H.323 Gatekeeper Default = Off
Enable
This settings enables gatekeeper operation.
H.323 Signaling over Default = Disabled. For hosted deployments, default = Preferred.
TLS
When enabled, TLS is used to secure the registration and call signaling
communication between IP Office and endpoints that support TLS. The H.323 phones
that support TLS are 9608, 9611, 9621, and 9641 running firmware version 6.6 or
higher.
When enabled, certificate information is configured in the 46xxSettings.txt file
on IP Office and automatically downloaded to the phone. When IP Office receives
a request from the phone for an identity certificate, IP Office searches its trusted
certificate store and finds the root CA that issued its identity certificate. IP Office
then provides the root CA as an auto-generated certificate file named Root-CA-
xxxxxxxx.pem.
For information on IP Office certificates, see Security Mode | System | Certificates.
The options are:
• Disabled: TLS is not used.
• Preferred: Use TLS when connecting to a phone that supports TLS.
• Enforced: TLS must be used. If the phone does not support TLS, the connection is
rejected.
When set to Enforced, the Remote Call Signaling Port setting is disabled.
If TLS security is enabled (Enforced or Preferred), it is recommended that you
enable a matching level of media security on System | System | VoIP | VoIP
Security.
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Field Description
H.323 Remote Extn Default = Off.
Enable
The system can be configured to support remote H.323 extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office is
located behind a corporate NAT/Firewall router and/or the H.323 phone is located
behind residential NAT enable router.
The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling
H.323 Remote Extn Enable allows configuration of the RTP Port number Range
(NAT) settings.
Auto-create Extn Default = Off
The field to set up auto creation of extensions for H.323 phones registering
themselves with the System as their gatekeeper. If selected, the system displays
the Auto Create Extension Password window prompting you to type a Password
and Confirm Password. This password is used for subsequent auto creation of
extensions. A message H.323 Auto-Create Extension option is active
is flashed next to the Auto Create Extension field till the option is cleared. SIP
Extensions use a separate setting, see below. This setting is not supported on
systems configured to use WebLM server licensing.
If using resilience backup to support Avaya IP phones, Auto-create Extn and Auto-
create User should not be left enabled after initial configuration or any subsequent
addition of new extensions and users. Leaving auto-create options enabled on a
system that is a failover target may cause duplicate extension/user records on the
multi-site network under multiple failure scenarios.
For security, any auto-create settings that are enabled are automatically disabled
after 24 hours.

SIP Trunks Enable

Field Description
SIP Trunks Enable Default = On.
This settings enables support of SIP trunks. It also requires entry of SIP Trunk
Channels licenses.
Enabling SIP Trunks Enable allows configuration of the RTP Port number Range
(NAT) settings.

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SIP Registrar Enable

Field Description
SIP Registrar Enable Default = Off.
If enabled, the IP Office can act as a SIP Registrar to which SIP endpoints register.
• Separate SIP registrars can be configured on LAN1 and LAN2.
• Registration of a SIP endpoint requires an available IP Endpoints license.
• SIP endpoints are also still subject to the extension capacity limits of the system.
SIP Remote Extn Default = Off.
Enable
The system can be configured to support remote SIP extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office
is located behind a corporate NAT/Firewall router and/or the SIP phone is located
behind residential NAT enable router.
• This option cannot be enabled on both LAN1 and LAN2.
• The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling SIP
Remote Extn Enable allows configuration of:
• the Remote UDP Port, Remote TCP Port, Remote TLS Port settings
• the Port Number Range (NAT) settings
Allowed SIP User Default = Block Blacklist Only
agents
The drop-down menu to select which SIP devices are allowed to register with the IP
Office system. Depending on the selection, IP Office allows registration of SIP User
Agents specified using the System > VOIP > Access Control Lists tab. The options
are:
• Allow All: Do not block any devices based on the UI strings.
• Block Blacklist Only: Block devices whose UA string is listed in the SIP UA
Blacklist.
• Avaya Clients & Whitelisted: Only allow devices with an Avaya UA string or
whose UA string is listed in the SIP UA Whitelist.
• Avaya Clients Only: Only allow clients with an Avaya UA string.
• Whitelisted only: Only allow devices whose UA string is listed in the SIP UA
Whitelist.
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 LAN1

Field Description
Auto-create Extn/User Default = Off.
The field to set up auto creation of extensions for SIP phones registering themselves
with the SIP registrar. If selected, the system prompts you to enter and confirm the
password is used for subsequent auto creation of extensions.
• This setting is not supported on systems configured to use WebLM server licensing.
• For security, any auto-create settings set to On are automatically set to Off after 24
hours.
SIP Domain Name Default = Blank
This value is used by SIP endpoints for registration with the IP Office system.
SIP endpoints register with IP Office using their SIP address that consists of their
phone number and IP Office SIP domain. Since IP Office does not allow calls from
unauthorized entities, the SIP domain does not need to be resolvable. However, the
SIP domain should be associated with FQDN (Fully Qualified Domain Name) for
security purposes. The entry should match the domain suffix part of the SIP Registrar
FQDN below, for example, example.com. If the field is left blank, registration uses
the LAN 1, LAN2, or public IP address.

Note:
For Avaya SIP telephones supported for resilience, the SIP Domain Name must
be common to all systems providing resilience.
SIP Registrar FQDN Default = Blank
The fully-qualified domain name to which the SIP endpoint send their registration
requests. For example, sbc.example.com.
• This FQDN is also used for Avaya Cloud Services and Avaya Push Notification
Services
The customer DNS must resolve this FQDN to an IP address that routes to the IP
Office. That is:
• For local extensions, the IP address of the IP Office LAN.
• For remote extensions, the external IPv4 address of the Avaya SBC or customer
firewall that routes to the IP Office.
Challenge Expiry Time Default = 10.
(secs)
The challenge expiry time is used during SIP extension registration. When a device
registers, the IP Office SIP Registrar sends a challenges and waits for a response. If
a response is not received within this timeout, the registration fails.
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System

Field Description
Layer 4 Protocol Default = TCP 5060 + UDP 5060.
Sets the ports on which the IP Office listens for SIP extension connections. Note that
most SIP clients use TLS/TCP/UDP in order of priority unless configured otherwise,
and will not fallback to a lower priority protocol even if it is enabled on the IP Office.
• UDP Port: Default = 5060 Enabled.
• TCP Port: Default = 5060 Enabled.
• TLS Port: Default = 5061 Disabled.
The following additional port settings are used if SIP Remote Extn Enable is
selected. Otherwise, the ports above are used for all SIP extension connections.
They set the ports the ports on which the IP Office listens for SIP extension
connections from remote extensions:
• Remote UDP Port: Default = 5060 Enabled.
• Remote TCP Port: Default = 5060 Enabled.
• Remote TLS Port: Default = 5061 Disabled.

RTP
Field Description
Port Number Range For each VoIP call, a receive port for incoming Real Time Protocol (RTP) traffic is
selected from a defined range of possible ports, using the even numbers in that
range. The Real Time Control Protocol (RTCP) traffic for the same call uses the RTP
port number plus 1, that is the odd numbers.
On some installations, it may be a requirement to change or restrict the port range
used. It is recommended that only port numbers between 49152 and 65535 are used,
that being the range defined by the Internet Assigned Numbers Authority (IANA) for
dynamic usage.

Important:
The minimum and maximum settings of the port range should only be adjusted
after careful consideration of the customer network configuration and existing
port usage. The gap between the minimum and maximum port values must be at
least 254.
Port Range (minimum) Default: IP500 V2 = 46750/Linux = 40750. Range = 1024 to 65530.
This sets the lower limit for the RTP port numbers used by the system.
Port Range Default = 50750. Range = 1024 to 65530.
(maximum)
This sets the upper limit for the RTP port numbers used by the system.

Port Number Range (NAT)

These settings are available when either H.323 Remote Extn Enable, SIP Trunks Enable, or SIP
Remote Extn Enable is set to On.
This option is not supported if System | LAN | Network Topology | Firewall/NAT Type is set to
Symmetric Firewall or Open Internet.

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Field Description
Port Range (minimum) Default: IP500 V2 = 46750/Linux = 40750. Range = 1024 to 65530.
This sets the lower limit for the RTP port numbers used by the system.
Port Range Default = 50750. Range = 1024 to 65530.
(maximum)
This sets the upper limit for the RTP port numbers used by the system.
Enable RTCP Monitor Default = On.
On Port 5005
For 1600, 4600, 5600, 9600 and J100 Series phones, the system can collect VoIP
QoS (Quality of Service) data from the phones. For other phones, including non-IP
phones, it can collect QoS data for calls that use a VCM channel. The QoS data
collected by the system is displayed by the System Status Application.
• This setting is mergeable. However, it is only applied to IP phones when they
register with the system. Therefore, any change to this setting requires the IP
phones that have already registered to be rebooted. IP phones can be remotely
rebooted using the System Status Application.
• The QoS data collected includes: RTP IP Address, Codec, Connection Type,
Round Trip Delay, Receive Jitter, Receive Packet Loss.
• This setting is not the same as the RTCPMON option within Avaya H.323 phone
settings. The system does not support the RTCPMON option.
RTCP collector IP Default = Blank.
address for phones
Sets the destination for the RTCP Monitor data described above. This enables you to
send the data collected to a third party QoS monitoring application.
The Enable RTCP Monitor On Port 5005 must be turned Off to enable this field.
Changes to this setting requires a reboot of the phones.

Keepalives
These settings are used to keep open external connections through devices such as firewalls and
session-border controllers. You can use these settings when the IP Office has connections to SIP
trunks and/or H323 and SIP remote workers.
Field Description
Scope Default = Disabled
Select whether the sending of keepalive packets should be disabled or sent for RTP
or for both RTP and RTCP.
Periodic timeout Default = 0 (Off). Range = 0 to 180 seconds.
Sets how long the system will wait before sending a keepalive if no other packets of
the select SCOPE are seen.
Initial keepalives Default = Disabled.
If enabled, keepalives can also been sent during the initial connection setup.

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System

DiffServ Settings
When transporting VoIP over low speed links, data packets (1500 byte packets) can block or delay
voice packets (typically 67 or 31 bytes). This can cause poor speech quality. Therefore, all traffic
routers in a network should support Quality of Service (QoS).
The IP Office system supports the DiffServ (RFC2474) QoS mechanism. This uses a Type of
Service (ToS) field in the IP packet header.
The IP Office applies the LANs DiffServ settings to outgoing traffic on any SIP lines which have
Line | SIP Line | Transport | Use Network Topology Info set to match the LAN interface.
• The hex and decimal entry fields for the following values are linked. The hex value is equal to
the decimal multiplied by 4.
• Do not use the same values for call signaling and call media (audio and voice).
• For correct operation, the same value must be set at both ends.
Field Description
DSCP (Hex) Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DiffServ Code Point (DSCP) setting applied to the media on VoIP calls. By
default, this value is applied to both audio and video unless a separate video value is
set.
Video DSCP (Hex) Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DSCP setting applied to video VoIP calls.
DSCP Mask (Hex) Default = FC (Hex)/63 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The mask applied to packets for the DSCP value.
SIG DSCP (Hex) Default = 88 (Hex)/34 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
This DSCP setting applied to the call signaling on VoIP calls. This must not match the
settings used for the media.

DHCP Settings
Field Description
Primary Site Specific Default = 176. Range = 128 to 254.
Option Number
A site specific option number (SSON) is used as part of DHCP to request additional
(4600/5600)
information. 176 is the default SSON used by 4600 Series and 5600 Series IP
phones.
Secondary Site Default = 242. Range = 128 to 254.
Specific Option
Similar to the primary SSON. 242 is the default SSON used by 1600 and 9600 Series
Number (1600/9600)
IP phones requesting installation settings via DHCP.
VLAN Default = Not present. This option is applied to H.323 phones using the system for
DHCP support. If set to Disabled, the L2Q value indicated to phones in the DHCP
response is 2 (disabled). If set to Not Present, no L2Q value is included in the DHCP
response.
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Field Description
1100 Voice VLAN Default = 232.
Site Specific Option
This is the SSON used for responses to 1100/1200 Series phones using the system
Number (SSON)
for DHCP.
1100 Voice VLAN IDs Default = Blank.
For 1100/1200 phone being supported by DHCP, this field sets the VLAN ID that
should be provided if necessary. Multiple IDs (up to 10) can be added, each
separated by a + sign.

Related links
LAN1 on page 213

Network Topology
Navigation: System | LAN | Network Topology
These settings are used for support of external SIP trunks when not using an SBC. They are also
used for supporting remote SIP/H323 extensions.
Network Address Translation (NAT) Overview
The network address translation (NAT) done by firewalls can affect VoIP calls. Two methods that
can be used to overcome this are STUN or TURN.
NAT Method Description
STUN STUN ("Session Traversal for NAT") is a mechanism to overcome the effect of some NAT
firewalls. In summary:
• The device configured for STUN sends test packets to the STUN server address.
These go through the firewall NAT process.
• The STUN server replies, including in the reply copies of the original packets it
received.
• By comparing the packets sent and received, the sender can try to determine the type
of NAT applied. It can then modify future packets it sends to other destinations to
overcome the effects of the firewall NAT.
TURN TURN ("Traversal Using Relays around NAT") is a NAT traversal mechanism that works
by relaying all traffic via a TURN server. This is typically a TURN service provided by the
customer's SBC.

STUN allows direct connection between the sender and receiver once setup, but is more restricted
in the types of NAT with which it can work. TURN supports more types of NAT, but also needs
to relay all traffic between the sender and receiver via the TURN server. STUN is easier to
implement and maintain compared to TURN, however most SBC devices support TURN.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
General
These settings are used by the IP Office for connection to a STUN server to support SIP trunks.

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Field Description
IP Office STUN Server Default = Blank
The IP address or fully qualified domain name (FQDN) of the STUN server the IP
Office should use. The system will send basic SIP messages to this destination and
from data inserted into the replies can try to determine the type NAT changes being
applied by any firewall between it and the ITSP.
Port Default = 3478.
Sets the port to which the STUN requests are sent.
Run STUN This button tests STUN operation between the system LAN using the settings
above. The results are used to automatically fill the NAT fields with appropriate
values discovered by the system. A information icon is then shown against the
fields to indicate that the values were automatically discovered rather than manually
entered.
Before using Run STUN, the SIP trunk must be configured.
Run STUN on startup Default = Off
This option is used in conjunction with values automatically discovered using Run
STUN. When selected, the system reruns STUN discovery whenever the system is
rebooted or connection failure to the SIP server occurs.

WebRTC
These settings are used for remote User Portal users using WebRTC (Softphone mode) to make
and receive calls using STUN and/or TURN. The values set are provided to the remote user portal
sessions through their normal MTCTI connection.
Field Description
WebRTC Client STUN Default = Blank (use stun.freeswitch.org:3478)
Server
Set the IP address or FQDN of the STUN server that the clients should use.
Port Default = 3748
The port the clients should use for STUN.
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 LAN1

Field Description
WebRTC Client Turn Default = Blank
Server
This is used for solutions that use a TURN service configured on an SBC. It provides
the IP address or FQDN of the TURN service.
• You can add the required port by adding :<port number>. For example add
:3748 to the address or FQDN.
• You can set the required transport method by adding ?transport=udp or ?
transport=tcp to the address or FQDN. By default UDP is assumed.
• The TURN server connection uses the name and password of an IP Office service
user.
- The service user must be a member of the security rights group TURN Server
with TURN Server Connection enabled.
- On new and defaulted systems, a service user called TURNServer exists and
is a member of the TURN Server rights group. However the service user is
disabled by default.
• The details of the TURN server address, name and password are passed to IP
Office User Portal sessions using their MTCTI connection to the IP Office.

NAT
The following fields can be completed either manually or the system can attempt to automatically
discover the appropriate values using Run STUN.
To complete the fields automatically:
1. Check that the SIP trunk to the ITSP is configured.
2. Set the IP Office STUN Server address.
3. Test STUN by clicking Run STUN.
4. Close and reload the configuration. If STUN was successful, the remaining fields are
updated using the results. A icon is shown against the fields to indicate that the values
were automatically discovered rather than manually entered.
Field Description
Firewall/NAT Type Default = Unknown
The settings here reflect different types of network firewalls. For descriptions of the
various options, see the table below.
Binding Refresh Time Default = 0 (Never). Range = 0 to 3600 seconds.
(seconds)
To keep the firewall port open for incoming calls, the system can send recurring SIP
OPTIONS requests to the remote proxy terminating the trunk. This setting configures
the frequency of those requests.
If you do not set a binding refresh time, you may experience problems receiving
inbound SIP calls after a short period of normal operation.
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System

Field Description
Public IP Address Default = 0.0.0.0
(IPv4)
If no address is set, the system's LAN1 address is used.
SIP Registrar public The public port values for UDP, TCP, and TLS.
ports
• UDP - Default = 5060
• TCP - Default = 5056
• TLS - Default = 5061

Firewall/NAT Type Description

Blocking Firewall –
Full Cone NAT A full cone NAT is one where:
• All requests from the same internal IP address and port are mapped to the same
external IP address and port.
• Any external host can send a packet to the internal host, by sending a packet to
the mapped external address.
• SIP packets need to be mapped to NAT address and Port.
• Any host in the internet can call on the open port. The local info in the SDP will
apply to multiple ITSP Hosts.
Open Internet If this mode is selected, the IP Office ignores settings obtained by STUN lookups.
The IP address used is that of the IP Office system's LAN interface.
One-To-One NAT This setting supports deployments where the IP Office is behind a NAT that performs
IP address translation but not port mappings. All required ports must be open on the
NAT.
When set to One-To-One NAT, the following configuration settings are applied and
cannot be edited.
• The NAT > SIP Registrar public ports values are set to 0.
• The LAN1 > VoIP > SIP Registrar Enable remote protocol port values are set to
equal their corresponding local protocol port values.
• The LAN1 > VoIP > RTP > Port Number Range (NAT) RTP | Port
Number Range (NAT) values are set to equal the corresponding Port Number
Range values.
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 LAN1

Firewall/NAT Type Description

Port Restricted Cone Similar to a Restricted Cone NAT, but the restriction includes port numbers.
NAT Specifically, an external host can send a packet, with source IP address X and
source port P, to the internal host only if the internal host had previously sent a
packet to IP address X and port P. SIP packets needs to be mapped. Keep-alives
must be sent to all ports that will be the source of a packet for each ITSP host IP
address. If this type of NAT/Firewall is detected or manually selected, no warning will
be displayed for this type of NAT.
Some Port Restricted NAT's have been found to be more symmetric in behavior,
creating a separate binding for each opened Port, if this is the case the manager
will display a warning ‘Communication is not possible unless the STUN
server is supported on same IP address as the ITSP’ as part of the
manager validation.
Restricted Cone NAT A restricted cone NAT is one where all requests from the same internal IP address
and port are mapped to the same external IP address and port. Unlike a full cone
NAT, an external host (with IP address X) can send a packet to the internal host only
if the internal host had previously sent a packet to IP address X. SIP packets needs
to be mapped. Responses from hosts are restricted to those that a packet has been
sent to. So if multiple ITSP hosts are to be supported, a keep alive will need to be
sent to each host. If this type of NAT/Firewall is detected or manually selected, no
warning will be displayed for this type of NAT.
Static Port Block Use the RTP Port Number Range specified on the VoIP tab without STUN
translation. Those ports must be fixed as open on any NAT firewall involved
Symmetric Firewall SIP packets are unchanged but ports need to be opened and kept open with keep-
alives.
• If this type of NAT is detected or manually selected, a warning ‘Communication
is not possible unless the STUN server is supported on same
IP address as the ITSP’ is displayed as part of the manager validation.
Symmetric NAT A symmetric NAT is one where all requests from the same internal IP address and
port, to a specific destination IP address and port, are mapped to the same external
IP address and port. If the same host sends a packet with the same source address
and port, but to a different destination, a different mapping is used. Furthermore,
only the external host that receives a packet can send a UDP packet back to the
internal host. SIP Packets need to be mapped but STUN will not provide the correct
information unless the IP address on the STUN server is the same as the ITSP
Host.
• If this type of NAT is detected or manually selected, a warning ‘Communication
is not possible unless the STUN server is supported on same
IP address as the ITSP’ is displayed as part of the manager validation.
Unknown The type of NAT is unknown or could not be determined.

SBC
These settings are used to provide values to remote extensions that connect to the IP Office
through an ASBCE. The values set are passed to the phones using methods that vary
depending on the phone type. For example, by altering the values in the auto-generated
46xxsettings.txt file when requested by a remote phone.

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System

These settings replace the RW_SB... NoUser source numbers used in pre-R11.1.2.4 systems,
which should be removed once replaced with these values.
Field Description
Public IP Address Default = Blank
(IPv4)
The public IPv4 address that routes to the public/external side of the ASBCE.
Depending on the customer network, this can be the public IP address of another
device such as a firewall that forwards to the SBC.
Public IP Address Default = Blank
(IPv6)
As above but using an IPv6 address. Use of an IPV6 address is supported for:
• Avaya Workplace Client R3.35 (Android and iOS).
• IP Office R11.1.3.1 or higher.
• ASBCE 10.1.2 or higher.
For further information, see the Deploying Remote IP Office SIP Phones with an
ASBCE manual.
Private IP Address Default = Blank
(IPv4)
The private IPv4 address of the ASBCE.
FQDN Default = Blank
The fully-qualified domain name of the ASBCE. You must set this value.
• The IP Office uses this value in the auto-generated 46xxsettings.txt file
requested by remote Avaya Workplace Client extensions. For other remote SIP
extensions, the IP Office uses the SIP Registrar FQDN.
• The customer DNS must resolve this FQDN to an IP address that routes to the IP
Office. That is:
- For remote extensions, the external IPv4 address of the Avaya SBC or customer
firewall that routes to the IP Office.
- If supporting remote Avaya Workplace Client extensions using IPv6, the FQDN
must resolve to both the external IPv4 and IPv6 addresses of the Avaya SBC or
customer firewall that routes to the IP Office.
SBC Registrar public The public ports on which the ASBCE is configured to listen for incoming SIP call.
ports
• UDP - Default = 5060
• TCP - Default = 5056
• TLS - Default = 5061

Related links
LAN1 on page 213

DHCP Pools
Navigation: System | LAN | DHCP Pools

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 LAN1

DHCP pools allows for the configuration of of IP address pools for allocation by the system when
acting as a DHCP server. On an IP500 V2 system, you can configure up to 8 pools. On Server
Edition Linux systems, you can configure up to 64 pools.
By default the DHCP settings (IP Address, IP Mask and Number of DHCP IP Addresses) set
on the LAN Settings tab are reflected by the first pool here. For support of PPP Dial In address
requests, at least one of the pools must be on the same subnet as the system's LAN. Only
addresses from a pool on the same subnet as the system's own LAN address will be used for PPP
Dial In.
These settings are mergeable. However, the following actions require a merge with service
disruption:
• Changing the Start Address, Subnet Mask or Default Router value for an existing DHCP
Pool of addresses.
• Decreasing Pool Size for an existing DHCP Pool of addresses.
• Deleting an existing DHCP Pool of addresses.
When these actions are performed, the DHCP (Server or DialIn) is re-initialized which triggers a
reboot of the Avaya DHCP Clients (H.323 and SIP) in order to force the Avaya DHCP clients to
renew their IP address lease and apply the new settings. For the remaining Avaya and non-Avaya
DHCP clients, you must manually reboot the devices in order to force the IP Addresses lease
renewal. Otherwise, the devices continue to use the allocated IP addresses until the IP addresses
lease time out expires. IP address lease time out is set to three days.
The DHCP server re-initialization causes a reboot of all Avaya DHCP clients and not only of the
DHCP clients that have obtained an IP Address within the modified DHCP Pool IP range. Note
that IP Office supports phone reboot only for E129 and B179 SIP phone models.
Field Description
Apply to Avaya IP Default = Off.
Phones Only
When set to On, the DHCP addresses are only used for requests from Avaya IP phones.
Other devices connected to the system LAN will have to use static addresses or obtain
their address from another DHCP server.
In addition to the above control, Avaya IP phones will only complete DHCP against
a DHCP server configured to supports a Site Specific Option Number (SSON) that
matches that set on the phone. The SSON numbers supported by the system DHCP are
set on the VoIP sub-tab.
Once set to On and the configuration has been merged, you must manually reboot the
non-Avaya DHCP Client devices in order to force IP addresses lease renewal and to
make the settings new values effective. Otherwise the non-Avaya DHCP Client devices
will continue to use the allocated IP addresses until the IP addresses lease time out
expires. IP address lease time out is set to three days.
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System

Field Description
DHCP Pool Up to 8 pools can be added. The first pool matches the IP Address, IP Mask and Number
of DHCP IP Addresses on the LAN Settings sub-tab. When adding or editing pools,
Manager will attempt to warn about overlaps and conflicts between pools. The options
are:
• Start Address Sets the first address in the pool.
• Subnet Mask: Default = 255.255.255.0 Sets the subnet mask for addresses issued
from the pool.
• Default Router: Default = 0.0.0.0 For pools issuing IP addresses on the same subnet
as the system LAN's, 0.0.0.0 instructs the system to determined the actual default
router address to issue by matching the IP address/subnet mask being issued in the
IP Routing table. This matches the default behaviour used by systems without multiple
pools. For pools issuing addresses not on the same subnet as the system LAN's, the
default router should be set to the correct value for devices on that subnet.
• Pool Size: Default = 0 Set the number of DHCP client addresses available in the pool.

Related links
LAN1 on page 213

LAN2
Navigation: System | LAN2
These settings used to configure the system's second LAN interface. The fields available for LAN2
are the same as for LAN1 except for the following additional field.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Firewall Default = <None> (No firewall)
Allows the selection of a system firewall to be applied to traffic routed from LAN2 to
LAN1.

Related links
System on page 203

DNS
Navigation: System | DNS

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 DNS

These settings configure the servers to which the IP Office system should send request when it
needs to resolve name addresses into numeric IP addresses.
• DNS is a mechanism through which the URL's such as www.avaya.com are resolved into
IP addresses. Typically the customer's internet service provider (ISP) specifies the address
of the DNS server their customers should use. In more complex networks, the customer may
host their own DNS server.
• WINS (Windows Internet Name Service) is a mechanism used within a Windows network to
convert PC and server names to IP addresses using a WINS server.
If the IP Office system is acting as a DHCP server, in addition to providing clients with their own
IP address settings, it can also provide them with their DNS and WINS settings if requested by the
client.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Configuration Settings
Field Description
DNS Service IP Address Default = 0.0.0.0 (Do not provide DNS/Use DNS forwarding)
This is the IP address of a DNS Server. If this field is left blank, the system uses
its own address as the DNS server for DHCP client and forward DNS requests
to the service provider when Request DNS is selected in the service being used
(Service > IP).
The IP Office does not support DNS priority. If the DNS response contains multiple
addresses with priority, the IP Office only uses the first address.
Backup DNS Server IP Default = 0.0.0.0 (No backup)
Address
This is an alternate DNS server address used in the server address above does not
respond.
DNS Domain Default = Blank (No domain)
This is the domain name for your IP address. Your Internet service provider or
network administrator provides this. Typically this field is left blank.
WINS Server IP Default = 0.0.0.0 (Do not provide WINS)
Address
This is the IP address of your local WINS server. This is only used by Windows
PCs, and normally points to an NT server nominated by your network administrator
as your WINS server. Setting a value will result in also sending a mode of "hybrid".
For Server Edition this field is only available on Expansion System (V2) servers.
Backup WINS Server IP Default = 0.0.0.0 (No backup)
Address
This is alternate WINS server address used if the server address above does not
respond.
WINS Scope Default = Blank (no scope)
This is provided by your network administrator or left blank. For Server Edition
systems, this field is only available on Expansion System (V2) servers.

Related links
System on page 203

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System

Voicemail
Navigation: System | Voicemail
Additional configuration information
For information on configuring Voicemail Pro resiliency, see Server Edition Resiliency on
page 806.
Configuration settings
The following settings are used to set the system's voicemail server type and location. Fields are
enabled or grayed out as appropriate to the selected voicemail type. Refer to the appropriate
voicemail installation manual for full details.
These settings are mergeable with the exception of Voicemail Type and Voicemail IP Address.
Changes to these settings requires a reboot of the system.
Voicemail Type
Field Description
Voicemail Type
Sets the type of voicemail service used by the IP Office server.
None No voicemail operation.
Analogue Trunk Select this option to support receiving a message waiting indicator (MWI) signal from
MWI analog trunks terminating on the ATM4U-V2 card. MWI is a telephone feature that turns
on a visual indicator on a telephone when there are recorded messages.
Avaya Aura Select this option if you want to configure the system to use Avaya Aura Messaging
Messaging as the central voicemail system. If you choose this option, you are still able to
use Embedded Voicemail or Voicemail Pro at each branch to provide auto-attendant
operation and announcements for waiting calls. When selected, access to voicemail is
routed via an SM line to the numbers specified in the AAM Number field. The optional
AAM PSTN Number can be configured for use when the SM Line is not in service.
For a setup where the voicemail box numbers configured on Avaya Aura Messaging
or Modular Messaging are same as the caller's DID, the short code to route the PSTN
call should be such that the caller-id is withheld ( "W" in the telephone-number of the
shortcode ). This is to make sure that, during rainy day - the voicemail system does not
automatically go to the voicemail box of the caller based on the caller id.
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 Voicemail

Field Description
Call Pilot Select this option if you want to configure the system to use CallPilot over SIP as the
central voicemail system. If you choose this option, you are still able to use Embedded
Voicemail or Voicemail Pro at each branch to provide auto-attendant operation and
announcements for waiting calls. When selected, access to voicemail is routed via SM
line to the numbers specified in the CallPilot Number field.
• The CallPilot PSTN Number field and associated Enable Voicemail Instructions
Using DTMF check box are not supported. IP Office cannot access the CallPilot
system over the PSTN when the Session Manager line is down.
• Users can access their CallPilot voicemail by dialing the Voicemail Collect short code.
Access to CallPilot voicemail from Auto Attendant cannot be enabled by setting a
Normal Transfer action to point to the Voicemail Collect short code. If desired, it can be
enabled by setting a Normal Transfer action to point to the CallPilot number.
Centralized Select this option when using a Voicemail Pro system installed and licensed on another
Voicemail system in a multi-site network. The outgoing line group of the IP Office line connection to
the system with the Voicemail Pro is entered as the Voicemail Destination.
In a Server Edition network this option is used on the Secondary Server and expansion
systems to indicate that they use the Primary Server for as their voicemail server.
Distributed This option can be used when additional Voicemail Pro voicemail servers are installed in
Voicemail a SCN network and configured to exchange messages with the central voicemail server
using email. This option is used if this system should use one of the additional servers
for its voicemail services rather than the central server. This option is not supported by
Server Edition systems.
When selected:
• The Voicemail Destination field is used for the outgoing H.323 IP line to the central
system.
• The Voicemail IP Address is used for the IP address of the distributed voicemail
server the system should use.
Embedded IP500 V2 systems can store voicemail messages and prompts on the system's own
Voicemail memory card. It also supports internal auto-attendant configuration. For details, refer to
IP Office Embedded Voicemail Installation.
Group Voicemail This option is used to support third-party voicemail systems attached by extension ports
in the group specified as the Voicemail Destination. Not supported by Server Edition
systems.
Modular Select this option if you want to configure the system to use Modular Messaging over SIP
Messaging over as the central voicemail system.
SIP
• When selected, access to voicemail is routed via an SM line to the numbers specified
in the MM Number field.
• The optional MM PSTN Number can be configured for use when the SM Line is not in
service.
Remote Audix Select this option if using a remote Avaya Intuity Audix or MultiMessage voicemail
Voicemail system. Requires entry of an Audix Voicemail license. This option is not supported
by Server Edition systems.
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System

Field Description
Voicemail Lite/Pro Select this option when using Voicemail Pro. The IP address of the PC being used
should be set as the Voicemail IP Address. In a Server Edition network this option
is used on the Primary Server. It can also be used on the Secondary Server if the
Secondary server includes its own voice mail server. Use of Voicemail Pro requires
licenses for the number of simultaneous calls to be supported.

Field Description
Voicemail Mode Default = IP Office Mode.
This field is only shown here for Embedded Voicemail. For systems using Voicemail
Pro, it can be changed using the Default Telephony Interface setting shown in IP Office
Web Manager and the Voicemail Pro client.
Voicemail provided by the IP Office system can use either IP Office Mode or Intuity
Mode key presses for mailbox functions. End users should be provided with the
appropriate mailbox user guide for the mode selected. You can switch between modes
without losing user data, such as passwords, greetings, or messages.
The following user guides are available from the Avaya support web site:
• Using IP Office Embedded Voicemail Intuity Mode
• Using IP Office Embedded Voicemail IP Office Mode
• Using a Voicemail Pro Intuity Mode Mailbox
• Using a Voicemail Pro IP Office Mode Mailbox
Voicemail Defaults: Non-Server Edition = Blank, Server Edition = IP trunk connection to the Primary
Destination Server.
• When the Voicemail Type is set to Remote Audix Voicemail, Centralized Voicemail
or Distributed Voicemail, this setting is used to enter the outgoing line group of the
line configured for connection to the phone system hosting the central voicemail server.
• When the Voicemail Type is set to Group Voicemail, this setting is used to specify the
group whose user extensions are connected to the 3rd party voicemail system.
• When the Voicemail Type is set to Analogue Trunk MWI, this setting is used to
specify the phone number of the message center. All analogue trunks configured for
Analogue Trunk MWI must have the same destination.
Voicemail IP Defaults: Non-Server Edition = 255.255.255.255, Primary Server = Primary Server IP
Address Address.
This setting is used when the Voicemail Type is set to Voicemail Pro or Distributed
Voicemail. It is the IP address of the PC running the voicemail server that the system
should use for its voicemail services.
If set as 255.255.255.255, the control unit broadcasts on the LAN for a response from
a voicemail server. If set to a specific IP address, the system connects only to the
voicemail server running at that address.
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Field Description
Backup Voicemail Defaults: Primary Server = Secondary Server IP Address, All others = 0.0.0.0 (Off).
IP Address
This option is supported with Voicemail Pro. An additional voicemail server can be setup
but left unused. If contact to the voicemail server specified by the Voicemail IP Address
is lost, responsibility for voicemail services is temporarily transferred to this backup
server address.
Maximum Record Default = 120 seconds. Range = 30 to 180 seconds. This field is only available when
Time Embedded Voicemail is selected as the Voicemail Type. The value sets the maximum
record time for messages and prompts.
Messages Button Default = On.
Goes to Visual
Visual Voice allows phone users to check their voicemail mailboxes and perform action
Voice
such as play, delete and forward messages through menus displayed on their phone. By
default, on phones with a MESSAGES button, the navigation is via spoken prompts. This
option allows that to be replaced by Visual Voice on phones that support Visual Voice
menus. For further details see the button action.
Enable Outcalling Default = Off (Outcalling not allowed).
This setting is used to enable or disable system support for outcalling on Embedded
Voicemail and Voicemail Pro. When not selected, all outcalling and configuration of
outcalling through mailboxes is disabled. For Voicemail Pro, outcalling can also be
disabled at the individual user mailbox level using the Voicemail Pro client.

Voicemail Channel Reservations

These settings allow the channels used for calls into voicemail to be reserved for particular
functions. Unreserved channels can be used for any function but reserved channels cannot be
used for any function other than that indicated.
Field Description
Unreserved Default = All channels
Channels
This setting shows the number of voicemail channels, out of the total available, that have
not been reserved.
Auto-Attendant Default = 0
This setting sets the number of channels reserved for calls directed to one of the
configured auto-attendants. .
Announcements Default = 0
This setting sets the number of channels reserved for announcements. When no
channels are available, calls continue without announcements.
Voice Recording Default = 0
This setting sets the number of channels reserved for voice recording other than
mandatory voice recording (see below). If no channels are available, recording does
not occur though recording progress may be indicated.
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Field Description
Mailbox Access Default = 0
This setting sets the number of channels reserved for users accessing mailboxes to
collect messages.
Mandatory Voice Default = 0
Recording
This setting sets the number of channels reserved for mandatory voice recording. When
no channels are available for a call set to mandatory recording, the call is barred and the
caller hears busy tone.

Call Recording
These settings apply to call recording provided by Voicemail Pro.
Field Description
Maximum Default = 30 days. Range 1 to 365 days.
Recording
Used for subscription systems using Centralized Media Manager to store call recordings.
Retention (Days)
This field sets how long recordings should be kept in the recording library before it is
automatically deleted.
Auto Restart Default = 15 seconds
Paused Recording
The value used to set a delay after which recording is automatically resumed.
(sec)
Hide Auto Default = Cleared
Recording
In addition to the audible advice of call recording prompt, Avaya Workplace Client
displays a message that states the meeting or call is being recorded.
Play Advice on Call Default = On
Recording
Sets whether an advice warning is played to all callers when their call is being recorded.
It is a legal requirement in some countries to inform the callers before recording their
calls, therefore you must get confirmation before you turn this option off.
This option is not shown in IP Office Manager. It can be set through either IP Office Web
Manager or the Voicemail Pro client.

Speech AI
These settings as available on subscription mode systems. If enabled, the system can use text-to-
speech (TTS) and automatic speech recognition (ASR) services with auto-attendants and system
meet-me conferences.
Field Description
Google Speech AI Default = Off
If enabled, the system can use text-to-speech (TTS) and automatic speech recognition
(ASR) services with auto-attendants and system meet-me conferences.
Speech Language Default = Match the system locale language if possible.
Sets the default language used for TTS prompts. This can be overridden by the particular
setting of the auto-attendant or system meet-me conference.
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Field Description
Speech Voice Sets the voice to be used with the speech language. The number of voices available
varies depending on the speech language selected.

DTMF Breakout
Allows system defaults to be set. These are then applied to all user mailboxes unless the user's
own settings differ.
The Park & Page feature is supported when the system voicemail type is configured as
Embedded Voicemail or Voicemail Pro. It allows a call to be parked while a page is made to a
hunt group or extension. This feature can be configured for Breakout DTMF 0, Breakout DTMF 2,
or Breakout DTMF 3.
Park & Page is also supported on systems where Avaya Aura Messaging, Modular Messaging
over SIP, or CallPilot (for IP Office Aura Edition with CS 1000 deployments) is configured as
the central voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto
attendant operation.
Field Description
Reception/ The number to which a caller is transferred if they press 0while listening to the mailbox
Breakout (DTMF 0) greeting rather than leaving a message (*0 on Embedded Voicemail in IP Office Mode).
For voicemail systems set to Intuity emulation mode, the mailbox owner can also access
this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave
Mail action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success results
connections depending on whether the caller pressed 0 before or after the record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0) setting.
• When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
- Paging Number: Displays a list of hunt groups and users (extensions). Select a hunt
group or extension to configure this option.
- Retries: The range is 0 to 5. The default setting is 0.
- Retry Timeout: Provided in the format M:SS (minute:seconds). The range can be
set in 15-second increments. The minimum setting is 15 seconds and the maximum
setting is 5 minutes. The default setting is 15 seconds
Breakout (DTMF 2) The number to which a caller is transferred if they press 2while listening to the mailbox
greeting rather than leaving a message (*2 on Embedded Voicemail in IP Office Mode).
Breakout (DTMF 3) The number to which a caller is transferred if they press 3while listening to the mailbox
greeting rather than leaving a message (*3 on Embedded Voicemail in IP Office Mode).

Voicemail Code Complexity

Defines the requirements for the voicemail code.

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For IP Office systems that have Voicemail Type set to Centralized, the Voicemail Code
Complexity settings must be the same as the IP Office system that is connected to Voicemail
Pro.
Field Description
Enforcement Default = On.
When on, a user PIN is required. The enforcement is not forced during upgrade but after
checking, it can not be cleared.
Minimum Length Default = 6. Maximum 31 digits. Older configurations can continue to have 4 digits with a
maximum of 20 digits.
Complexity Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (111111).
• A PIN consisting of a sequence, forward or reverse, is not allowed (123456, 564321).
The number of users having invalid Voicemail Code complexity is highlighted below this
field in red colored text.

SIP Settings
For Embedded Voicemail and Voicemail Pro, for calls made or received on a SIP line where any
of the line's SIP URI fields are set to Use Internal Data, that data is taken from these settings.
These options are shown if the system has SIP trunks and is set to use Embedded Voicemail,
Voicemail Lite/Pro, Centralized Voicemail or Distributed Voicemail.
Field Description
SIP Name Default = Blank on Voicemail tab/Extension number on other tabs.
This value is used for fields, other the Contact header, where the SIP URI entry being
used has its Contact field set to Use Internal Data.
• On incoming calls, if the Local URI is set to Use Internal Data, the system can
potentially match the received R-URI or From header value to a user and/or group SIP
Name. This requires the SIP URIs Incoming Group to match a Incoming Call Route
with the same Line Group ID and a . (period) destination.
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display field of the SIP URI being used is set to
Use Internal Data.
Contact Default = Blank on Voicemail tab/Extension number on other tabs.
The value is used for the Contact header when the Contact field of the SIP URI being
used for a SIP call is set to Use Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs.
If the From field in the SIP URI is set to Use Internal Data, selecting this option inserts
Anonymous into that field rather than the SIP Name set above. See Anonymous SIP
Calls on page 842.

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Voicemail Language Prompts

When the system routes a call to the voicemail server it indicates the locale for which matching
prompts should be provided if available. The locale sent to the voicemail server by the system
is determined as show below. If the required set of prompts is not available, the voicemail will
fallback to another appropriate language and finally to English (refer to the appropriate voicemail
installation manual for details).
• Short Code Locale: The short code locale, if set, is used if the call is routed to voicemail
using the short code.
• Incoming Call Route Locale: The incoming call route locale, if set, is used if caller is
external.
• User Locale: The user locale, if set, is used if the caller is internal.
• System Locale: If no user or incoming call route locale is set, the system locale is used
unless overridden by a short code locale.
Systems using Embedded Voicemail, if the required set of upgraded language prompts to match
the locale is not present on the system SD card, Manager will display an error. The required
prompt set can be uploaded from Manager using the Add/Display VM Locales option.
Related links
System on page 203

Telephony
Used to set the default telephony operation of the system. Some settings shown here can be
overridden for individual users through their User | Telephony tab. The settings are split into a
number of sub-tabs.
Related links
System on page 203
Telephony on page 239
Park and Page on page 248
Tones and Music on page 249
Ring Tones on page 253
SM on page 253
MS Teams on page 254
Call Log on page 255
TUI on page 256

Telephony
Navigation: System | Telephony
Additional configuration information
• The Directory Overrides Barring setting allows you to control barred numbers. For additional
configuration information, see Call Barring on page 703.

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• The Inhibit Off-Switch Forward/Transfer stops any user from transferring or forwarding
calls externally. For additional information, see Off-Switch Transfer Restrictions on page 784.
• For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 621.
• For additional information on ring tones, see Ring Tones on page 655.
Configuration Settings
Used to configure a wide range of general purpose telephony settings for the whole system.
These settings are mergeable with the exception of Companding LAW and Media Connection
Preservation. Changes to these settings requires a reboot of the system.
Analog Extensions
These settings apply only to analog extension ports provided by the system. For Server Edition
this field is only available on Expansion System (V2) systems
Field Description
Default Outside Call Default = Normal. See Ring Tones on page 655.
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming external calls. For details of the ring types see System | Telephony | Ring
Tones.
This setting can be overridden by a user's User | Telephony | Call Settings setting.
Note that changing the pattern may cause fax and modem device extensions to not
recognize and answer calls.
Default Inside Call Default = Ring Type 1. See Ring Tones on page 655.
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming internal calls. For details of the ring types see System | Telephony | Ring
Tones. This setting can be overridden by a user's User | Telephony | Call Settings
setting.
Default Ring Back Default = Ring Type 2. See Ring Tones on page 655.
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
ringback calls such as hold return, park return, voicemail ringback, and Ring Back
when Free. For details of the ring types see System | Telephony | Ring Tones.
This setting can be overridden by a user's User | Telephony | Call Settings setting.
Restrict Analog Default = Off.
Extension Ringer
Supported on IP500 V2 systems only. If selected:
Voltage
• The ring voltage on analog extension ports on the system is limited to a maximum
of 40V Peak-Peak.
• The message waiting indication (MWI) settings for analog extension are limited to
Line Reversal A, Line Reversal B or None.
• Any analog extension already set to another MWI setting is forced to Line
Reversal A.

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Companding Law
Field Description
Companding Law These settings should not normally be changed from their defaults. They should only
be used where 4400 Series phones (ULAW) are installed on systems which have
A-Law digital trunks.
A-Law or U-Law> PCM (Pulse Code Modulation) is a method for encoding voice
as data. In telephony, two methods of PCM encoding are widely used, A-Law and
U-Law (also called Mu-Law or µ-Law). Typically U-Law is used in North America and
a few other locations while A-Law is used elsewhere. As well as setting the correct
PCM encoding for the region, the A-Law or U-Law setting of a system when it is first
started affects a wide range of regional defaults relating to line settings and other
values.
For IP500 V2 systems, the encoding default is set by the type of Feature Key
installed when the system is first started. The cards are either specifically A-Law or
U-Law.

Telephony
Field Description
Dial Delay Time (secs) Default = 4 (USA/Japan) or 1 (ROW). Range = 1 to 30 seconds.
This setting sets the time the system waits following a dialed digit before it starts
looking for a short code match. In situations where there are potential short codes
matches but not exact match, it also sets the delay following the dialing of a digit
before dialing complete is assumed.
Dial Delay Count Default = 0 digits (USA/Japan) or 4 digits (ROW). Range = 0 to 30 digits.
This setting sets the number of digits dialed after which the system starts looking for
a short code match regardless of the Dial Delay Time.
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Field Description
Default No Answer Default = 15 seconds. Range = 6 to 99999 seconds.
Time (secs)
This setting controls the amount of time before an alerting call is considered as
unanswered. How the call is treated when this time expires depends on the call type.
• For calls to a user:
• - the call follows the user's Forward on No Answer settings if enabled. If not set,
the call goes to voicemail if available or else continues to ring.
- This timer is also used to control the duration of call forwarding if the forward
destination does not answer.
- It also controls the duration of ringback call alerting.
- For a user, this setting is overridden by the user's User | Telephony | Call
Settings | No Answer Time setting if different.
• For calls to hunt groups:
- This setting controls the time before the call is presented to the next available
hunt group member.
- This setting is overridden by the group's Group | Fallback | Group No Answer
Time setting if different.
If the system includes users who are using Avaya Workplace Client on iOS devices,
it is recommended to set the time to at least 20 seconds. You should do this for
either the system default, or for the individual users and any hunt groups to which
they belong.
Hold Timeout (secs) Default = US: 120 seconds/ROW: 15 seconds. Range = 0 (Off) to 99999 seconds.
This setting controls how long calls remain on hold before recalling to the user who
held the call. The user's wrap-up time is also added.
Note that the recall only occurs if the user has no other connected call. Recalled
calls will continue ringing and do not follow forwards or go to voicemail.
Park Timeout (secs) Default = 300 seconds. Range 0 (Off) to 99999 seconds.
This setting controls how long calls remain parked before recalling to the user who
parked the call.
Note that the recall only occurs if the user has no other connected call. Recalled
calls will continue ringing and do not follow forwards or go to voicemail.
Ring Delay Default = 5 seconds. Range = 0 to 98 seconds.
This setting is used when any of the user's programmed appearance buttons is
set to Delayed ringing. Calls received on that button will initially only alert visually.
Audible alerting will only occur after the ring delay has expired.
This setting can be overridden by a ring delay set for an individual user (User |
Telephony | Multi-line Options | Ring Delay).
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Field Description
Call Priority Promotion Default = Disabled. Range = Disabled, 10 to 999 seconds.
Time (secs)
When calls are queued for a hunt group, higher priority calls are placed ahead of
lower priority calls, with calls of the same priority sort by time in queue. External calls
are assigned a priority (1-Low, 2-Medium or 3-High) by the Incoming Call Route
that routed the call. Internal calls are assigned a priority of 1-Low. This option can
be used to increase the priority of a call each time it has remained queued for longer
than this value. The calls priority is increased by 1 each time until it reaches 3-High.
In situations where calls are queued, high priority calls are placed before calls of a
lower priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where
Voicemail Pro is being used to provided queue ETA and queue position messages
to callers since those values will no longer be accurate when a higher priority
call is placed into the queue. Note also that Voicemail Pro will not allow a value
already announced to an existing caller to increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt
group's Queue Length Limit, the limit is temporarily raised by 1. This means that
calls already queued are not rerouted by the addition of a higher priority call into
the queue.
Default Currency Default = Locale specific.
This setting is used with ISDN Advice of Charge (AOC) services. Note that changing
the currency clears all call costs stored by the system except those already logged
through SMDR. The currency is displayed in the system SMDR output.
Default Name Priority Default = Favor Trunk.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting determines
which method is used by default. For each SIP line, this setting can be overridden by
the line's own Name Priority setting if required. Select one of the following options:
• Favor Trunk: Display the name provided by the trunk. For example, the trunk may
be configured to provide the calling number or the name of the caller. The system
should display the caller information as it is provided by the trunk. If the trunk does
not provide a name, the system uses the Favor Directory method.
• Favor Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
the name provided by the SIP line. If no match is found, the name provided by the
line, if any, is used.
Media Connection Default = Enabled.
Preservation
When enabled, attempts to maintain established calls despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
When enabled, Media Connection Preservation applies to SCN links and Avaya
H.323 phones that support connection preservation.
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Field Description
Phone Failback Default = Automatic.
Applies to H.323 phones that support resiliency. The options are:
• Automatic
• Manual
Phones are permitted to failover to the secondary gatekeeper when the IP Office
Line link to the primary gatekeeper is down.
When set to Automatic, if a phone’s primary gatekeeper has been up for more than
10 minutes, the system causes the phone to failback if the phone is not in use. If
the phone is in use, the system will reattempt failback 10 seconds after the phone
ceases to be in use.
When set to Manual, phones remain in failover until manually restarted or re-
registered, after which the phone attempts to fail back.

Note:
Manual failback is not supported on SIP phones.
DSS Status Default = Off
This setting affects Avaya display phones with programmable buttons. It controls
whether pressing a DSS key set to another user who has a call ringing will display
details of the caller. When off, no caller information is displayed.
Auto Hold Default = On (Off for the United States locale).
Used for users with multiple appearance buttons. When on, if a user presses
another appearance button during a call, their current call is placed on hold. When
off, if a users presses another appearance button during a call, their current call is
disconnected.
Show Account Code Default = On This setting controls the display and listing of system account codes.
• When on: When entering account codes through a phone, the account code digits
are shown while being dialed.
• When off: When entering account codes through a phone, the account code digits
are replaced by s characters on the display.
Inhibit Off-Switch Default = On
Forward/Transfer
When enabled, this setting stops any user from transferring or forwarding calls
externally.
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Field Description
Restrict Network Default = Off.
Interconnect
When this option is enabled, each trunk is provided with a Network Type option that
can be configured as either Public or Private. The system will not allow calls on
a public trunk to be connected to a private trunk and vice versa, returning number
unobtainable indication instead.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location Default = Off.
specific information
When set to On, this setting is available in the trunk configuration settings when
Network Type is set to Private.
Set to On if the PBX on the other end of the trunk is toll compliant.
Drop External Only Default = On.
Impromptu Conference
If selected, when the last remaining internal user in a conference exits the
conference, the conference is ended, regardless of whether it contains any external
callers.
If not selected, the conference is automatically ended when the last internal party or
trunk that supports reliable disconnect exits the conference. The Inhibit Off-Switch
Forward/Transfer option above is no longer applied to conference calls.
Visually Differentiate Default = Off.
External Call
This setting is applied to the lamp flashing rate used for bridged appearance and
call coverage appearance buttons on 1400, 1600 and 9600 Series phones and on
their button modules. When selected, external calls alerting on those buttons will use
a slow flash (200ms on/50ms off). If not selected or if the call is internal, normal
flashing (500ms on/500ms off) is used.
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Field Description
Unsupervised Analog Default = Off.
Trunk Disconnect
When using analog trunks, various methods are used for trunk supervision. That
Handling
is to detect when the far end of the trunk has disconnected and so disconnect the
local end of the call. Depending on the locale, the system uses Disconnect Clear
signaling and or Busy Tone Detection. This setting should only be enabled if it is
know that the analog trunks do not provide disconnect clear signaling or reliable
busy tone. For Server Edition this field is only available on Expansion System (V2)
systems.
When enabled:
• Disconnect Clear signaling detection is disabled. Busy tone detection remains on.
• Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not
allowed. The Allow Analog Trunk to Trunk Connect setting on analog trunks
(Line | Analog Options) is disabled.
• If Voicemail Pro is being used for external call transfers, Supervised Transfer
actions should be used in call flows rather than Transfer actions.
• All systems in the network must have this setting set to match each other.
High Quality Default = On.
Conferencing
Supports the use of the G.722 codec. IP lines and extensions using G.722 are
provided with wide band audio. If High Quality Conferencing is enabled, when
several wide band audio devices are in the same conference, the system will
ensure that the audio between them remains wide band, even if the conference also
contains other lines and devices using narrow band audio (analog devices, digital
devices and IP devices using codecs other than G.722).
Digital/Analogue Auto Default = On. (IP500 V2 only. Default = Off for Server Edition/On for others)
Create User
When enabled, an associated user is created for each digital/analogue extension
created. Digital/analogue extension creation occurs on initial start up, reset of
configuration, or addition of new digital/analogue expansion units or plug-in
modules.
Directory Overrides Default = On.
Barring
When enabled, barred numbers are not barred if the dialed number is in the External
Directory.
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Field Description
Advertize Callee State Default = Off.
To Internal Callers
When enabled, for internal calls, additional status information is communicated to
the calling party.
Not supported for SIP endpoints except for J100 Series phones (not including the
J129).
• When calling another internal phone and the called phone is set to Do Not Disturb
or on another call, the calling phone displays “Do Not Disturb” or “On Another Call”
rather than “Number Busy”.
• On 9500 Series, 9600 Series and J100 Series, if a line appearance is programmed
on a button on phone A and that line is in use on phone B, then phone A displays
the name of the current user of the line along with the line number.
• If a line appearance on a phone is in use elsewhere in the system and another
extension unsuccessfully attempts to seize that line, the phone displays “In
Use:<name>” where <name> is the name of the user currently using the line.
This configuration parameter sets the system wide default. Individual users can be
configured for this feature using the setting User | Telephony | Call Settings |
Advertize Callee State To Internal Callers
Internal Ring on Default = Off.
Transfer
When enabled, the transfer enquiry calls ring with internal ring tone even if the call
that is being transferred is an external call. If the user transferring the call completes
the call when the call is ringing, the ring tone played to the target changes to the ring
tone appropriate for the call being transferred.
This feature is supported on phone series: 1400, 9500, 1600, 9600, and analog
phones.
This feature is not supported on SIP and H.323 DECT phones.

Login Code Complexity

Defines the requirements for the login code.
Field Description
Enforcement Default = On.
When on, a user PIN is required.
Minimum Length Default = 6. Maximum 15 digits.
The number of users with login codes less than six digits is displayed below the field
in red colored text.
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Field Description
Complexity Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (111111).
• A PIN consisting of forward or backward sequence are not allowed. Examples:
123456, 654321.

RTCP Collector Configuration

Field Description
Send RTCP to an RTCP When the check box is selected, system RTCP reporting is enabled. For IP Office
Collector Release 10.0 and higher, in addition to having the individual phones send RTCP call
quality reports, the system can also send RTCP reports for calls.
Server Address This Sets the address of the third-party QoS monitoring application to which the
system sends RTCP reports.
UDP Port Number The destination port. The default for this filed is 5005.
RTCP reporting interval This setting sets the time interval at which the system sends RTCP reports.
(secs)

Related links
Telephony on page 239

Park and Page

Navigation: System | Telephony | Park and Page
The Park and Page tab allows for simple configuration of the of the short code and the
programmable button for the park and page function.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Central Park Range Default = Blank. Range = nX to nnnnnnnXX The park slot ID range definition, where n is
a digit sequence from 1 to 9999999 and X represents a park slot value from 0 to 99. The
Central Park Range cannot exceed 9 characters total length.
Examples:
• 1X defines range 10-19
• 3XX defines range 300-399
• 9876543XX defines range 987654300-987654399
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 Telephony

Field Description
Page Target Group Default = Blank. The list of paging group targets that are presented on supported phones
List if the Page action is requested after the Call Park.
On some phones, only the first three groups can be presented as Page options (via the
Softkeys on the phone). On phones that support scrolling lists, a larger list of possible
Page targets can be presented.

Related links
Telephony on page 239

Tones and Music

Navigation: System | Telephony | Tones and Music
Additional configuration information
For additional information on configuring hold music, see Music On Hold on page 657.
Configuration settings
Used to configure the various tones and music on hold sources used by the system.
The settings are mergeable except for Disconnect Tone and Busy Tone Detection. Changes to
these settings requires a reboot of the system.
Deleting any of the hold music Alternate Sources requires a reboot.
Field Description
Conferencing Tone Default = Entry & Exit Tones.
This settings controls how conference tones are used. The options are:
• Entry & Exit Tones
A single tone is heard when a new party joins a conference and double-tone is
heard when a party leaves the conference.
• Repeating Tone
A conference tone is heard every 10 seconds by all conference parties.
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System

Field Description
Disconnect Tone Default = Default (Use locale setting).
For digital and IP phones, when the system detects that the far end of a call has
disconnected, it can make the near end either go idle or play disconnect tone (analog
phones always play disconnect tone).
By default, the chosen behavior depends on the system locale. Note also that when
using disconnect tone, the tone used depends on the system locale.
• Default
Use the system locale default for disconnected calls. See Avaya IP Office Locale
Settings.
• On
Play disconnect tone when far end disconnection is detected.
• Off
Go idle when far end disconnection is detected.
Busy Tone Detection Default = Off.
Enables or disables the use of busy tone detection for call clearing. This is a system
wide setting.
CLI Type This field is used to set the CLI detection used for incoming analog trunks. Note that
the CLI Type field is shown for locales other than Customize.
For the Customize locale, it is set through the System | System form.
The options are DTMF, FSK V23 or FSK BELL202.
Local Dial Tone Default = On
For all normal operation this setting should be left enabled as it allows the system to
provide dial tone to users (essential for MSN working).
Local Busy Tone Default = Off
This setting should only be used when the local exchange gives a busy signal via
Q.931 but does not provide busy tone.
Beep on Listen Default = On
This setting controls whether call parties hear a repeating tone when their call is
monitored by another party using the Call Listen feature.

Warning:
• Listening to a call without the other parties being aware is subject to local
regulations. You must ensure that you have complied with the local regulations.
Failure to do so can result in penalties.
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 Telephony

Field Description
GSM Silence Default = Off.
Suppression
This setting should only be selected if voice quality problems are experienced with
calls to voicemail or while recording calls. When on, the system signals silence by
generating silence data packets in periods when the voicemail system is not playing
prompts. Note that use of this option may cause some timeout routing options in
voicemail to no longer work.
Analog Trunk VAD Default = Off.
Select this option to enable Voice Activity Detection (VAD) for analog trunks
terminating on the ATM4U-V2 card. VAD functionality provides a Call Answer signal
triggered by voice activity. This signal can be used for:
• Mobile Twinning
• SMDR
• Call Forwarding
• Call Display
• Mobile Call Control
• Transfer Ringing Call
• TAPI
• Trunk to Trunk Call
Busy Tone Detection Default = System Frequency (Defined by system locale. See Avaya IP Office Locale
Settings.)
Allows configuration of the system's busy tone detection settings on lines that do not
provide reliable disconnect signaling. In that case, the system will use tone disconnect
clearing to disconnect such lines after 6 seconds of continuous tone.
• The settings should only be adjusted if advised by Avaya Technical Support.
• Changes to this setting require a reboot when the new configuration is sent to the
system.
• For Server Edition, this field is only available on Expansion System (V2) systems.

Hold Music
This section is used to define the source for the system's music on hold source. You must ensure
that any MOH source you use complies with copyright, performing rights and other local and
national legal requirements.
Server Edition deployments support centralized music on hold, where the Primary Server streams
music to the Secondary Server and all expansion servers.
The WAV file properties must be:
• PCM, 8kHz 16-bit Mono.
• Maximum length: 90 seconds on IP500 V2 systems, 600 seconds on Linux-based systems.
If the file downloaded is in the incorrect format, it will be discarded from memory after the
download.

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Caution:
Copying files in the incorrect format directly into the opt/ipoffice/system/primary
directory can disable the music on hold function.
The WAV file used as the system source must be named HoldMusic.wav. For WAV files used as
alternate sources WAV files:
• Up to 27 IA5 characters with no spaces.
• Any file extension.
• On Linux-base systems, the filename is case sensitive.
Field Description
System Source Default = WAV File.
Selects the default hold music source. Note that changes to the System Source
requires a reboot. The options are:

Setting Description
WAV Use the HoldMusic.wav file. The IP Office loads the file
using TFTP, or you can directly add the file using the
embedded file manager.
WAV (restart) Identical to WAV except that for each new listener, the file
plays from the beginning.
• Not supported on IP500 V2 systems.
• Cannot be used as a centralized source.
External Applicable to IP500 V2 systems. Use the audio source
connected to the Audio port on the control unit.
Tone Use a double beep tone: 425Hz, 02./0.2/0.2/3.4 seconds on/
off.
• This tone is also used if the system source is set to WAV
File but the HoldMusic.wav file has not been successfully
loaded.
Alternate Sources You can assigned a configured alternate source as the Hold Music Source for an
Incoming Call Route or Group, overriding the default use of the system source. For
more details, see Alternate Source on page 659.
Adding and changing a source can be merged, but deleting a source requires a
reboot.
• Number: Automatically assigned by the system.
• Name: Up to 31 characters. Use this field to associate a name with the alternate
source. That name is then used to select the source in the Hold Music Source field
on Incoming Call Routes and Group settings.
• Source: Up to 31 characters. Defines the source for the music on hold.

Related links
Telephony on page 239

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 Telephony

Ring Tones
Navigation: System | Telephony | Ring Tones
Additional configuration information
For additional ring tone configuration information, see Ring Tones. on page 655
Configuration settings
Used to configure distinct ring tones for groups and incoming call routes. Ring tone override
features are only supported on 1400 Series, 9500 Series and J100 Series (except J129) phones.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Available Ring Tones In this table, the Number, Name, and Source values are system supplied. The Name
value is used to create a ring tone plan.
Ring Tone Plan Use this table to specify available ring tones. Ring tones in this table can be applied to
hunt groups and incoming call routes and by short codes.
• Number: System supplied.
The Number can be used in a short code by adding r(x) to the Telephone Number
field, where x = 1 to 8 and specifies which ring tone plan to use.
• Name: A descriptive name for where this ring tone is used. For example, the name
of a hunt group. Each name in the table must be unique. Once configured in this
table, ring tone names can be selected from the Ring Tone Override field at:
- Group | Group
- Incoming Call Route | Standard
• Ring Tone: The list of ring tone names from the Available Ring Tones table.

Related links
Telephony on page 239

SM
Navigation: System | Telephony | SM
Used to configure settings that apply to both SM lines.
These settings are not mergeable. Changes to these settings require a reboot of the system.

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System

Field Description
Branch Prefix Default = Blank. Maximum range = 15 digits.
This number is used to identify the IP Office system within the Avaya Aura® network.
On calls routed via an SM Line, the branch prefix is added as a prefix to the caller's
extension number.
• The branch prefix of each IP Office system must be unique and must not overlap.
For example 85, 861 and 862 are okay, but 86 and 861 overlap.
• You can leave the prefix blank. If you do not configure the branch prefix, the IP Office
user extensions must be defined with the user's full enterprise extension number.
Local Number Default = Blank (Off). Range = Blank or 3 to 9 in deployments with IP Office users and
Length blank or 3 to 15 in deployments with only centralized users.
This field sets the default length for extension numbers for extensions, users, and hunt
groups added to the IP Office configuration. Entry of an extension number of a different
length will cause an error warning.
The number of digits entered in the Branch Prefix field plus the value entered in
the Local Number Length field must not exceed 15 digits. You can leave the Local
Number Length field blank.
Proactive Monitoring Default = 60 seconds. Range = 60 seconds to 100000 seconds.
The branch IP Office system sends regular SIP OPTIONS messages to the SM line in
order to check the status of line. This setting controls the frequency of those messages
when the SM line is currently in service.
Monitoring Retries Default = 1. Range = 0 to 5.
The number of times the branch IP Office system retries sending an OPTIONS request
to Session Manager before the SM Line is marked out-of-service.
Reactive Monitoring Default 60 seconds. Range = 10 to 3600 seconds.
The branch IP Office system sends regular SIP OPTIONS messages to the SM line in
order to check the status of line. This setting controls the frequency of those messages
when the SM line is currently out-of-service.
User Shortcode Default = Rainy day.
Routing
Set when user dialing should be checked against IP Office user short codes and
processing of matches applied:
• Rainy day - Only check when no SM line connection is available.
• Always - Always check.

Related links
Telephony on page 239

MS Teams
Navigation: System > Telephony > MS Teams
These settings are applied to an IP Office system configured for MS-Teams direct routing. For
installation details, refer to the Deploying MS Teams Direct Routing with IP Office manual.

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 Telephony

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Auto Populate MS Default = Enabled.
Teams Data
When enabled, the user MS Teams URI settings cannot be edited. Instead, they are
controlled via the system's configured Azure Active Directory connection.

Related links
Telephony on page 239

Call Log
Navigation: System | Telephony | Call Log
The IP Office stores a centralized call log for each user:
• The user's centralized call log is stored by the IP Office system on which the user is
configured. If the user is logged in on another system, new call log records are sent to
the user's home IP Office system, but using the time and date on the IP Office system where
the user is logged in.
• Each user's centralized call log can contain 30 (IP500 V2) or 60 (Server Edition) call records.
Each new call record replaces the oldest previous record when it reaches the limit.
• By default, the centralized call log is displayed on Avaya phones with a fixed Call Log
or History button (1400, 1600, 9500, 9600, J100 Series) and in the one-X Portal, Avaya
Workplace Client, and IP Office User Portal applications.
• The centralized call log moves with the user as they log onto different phones and IP Office
applications.
• The missed call count is updated per unique caller, not per call.
These settings control the IP Office systems default application of user centralized call logs.
Additional user specific settings (User > Telephony > Call Log) can override and change the
system settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Default Centralized Default = On.
Call Log On
• If enabled, Avaya phones with a fixed Call Log or History button (1400, 1600, 9500,
9600, J100 Series) display the user's centralized call.
The use of centralized call logging can be enabled/disabled on a per user basis using
the setting User | Telephony | Call Log | Centralized Call Log.
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System

Field Description
Log Missed Calls Default = Off.
Answered at
This setting controls whether the users' centralized call logs should include calls that
Coverage
alert the users but are then answered elsewhere.
• Coverage applies for calls that alert the user and are then answered elsewhere. For
example by call pickup, call coverage buttons, a coverage group member, bridged
appearance button, user button, voicemail, and so on.
• Coverage does not apply for calls that do not alert the user and are answered
elsewhere. For example, forward unconditional.
• If covered by another user, that user will have an answered call in their centralized
call log.
Log Missed Hunt Default = Off.
Group Calls
This setting controls whether the user centralized call log can include missed hunt
group calls.
• If disabled, users' centralized call logs only include hunt group calls answered by the
user.
• If enabled, the IP Office also stores call logs of hunt group calls not answered by
anyone, including hunt group calls that go to voicemail.
- The IP Office system stores up to 10 call records for each hunt group. When this
limit is reached, new call records replace the oldest record.
- For each user, within the user call log settings (User | Telephony | Call Log), you
must select which hunt group's missed calls are displayed as part of the user's
centralized call log.

Related links
Telephony on page 239

TUI
Navigation: System | Telephony | TUI
Used to configure system wide telephony user interface (TUI) options for 1400, 1600, 9500, 9600
and J100 Series phones (except the J129).
Use these settings to define the default phone display when feature menus are disabled. Note that
for new users, the default phone display options are set to the system default values.
Feature menus can be disabled in one of two ways.
• Set System | Telephony | TUI | Features Menu to Off. Set User | Telephony | TUI | User
Setting to Same as System.
• On User | Telephony | TUI, set User Setting to Custom and set Features Menu to Off.
Configuration settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Telephony

Phone Type Variable Description

1400 Display Name Defines the default value of the User’s Features > Phone User >
Preference Phone Screen Settings > Display Name setting.
1600
Default = Off
When enabled, displays the user name.
9500 Column View Defines the default value of the User’s Features > Phone User >
Preference Phone Screen Settings > Display Mode setting.
9608
Default = Dual
9611
Column view can be Single or Dual.
9621 Quick Touch Panel Defines the default value of the User’s Features > Phone User >
Lines Phone Screen Settings > Quick Touch Lines setting.
9641
Default = Optimize
Sets the Quick Touch Panel number. The options are 1, 2, and
Optimize. When set to Optimize:
• 9621 = 1
• 9641 = 2

Field Description
Time Format Default = Locale Defined.
Set the system time format display. The default time format is defined by the Locale
setting. You can override the default and set the time format to a 12- hour or 24-hour
clock.
Features Menu Controls
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Field Description
Features Menu Default = On
When set to on, you can select to turn individual menus and features on users phone’s
on or off. The system level settings can be overridden at the individual user settings
level if required for particular users. The following feature menus are listed:
• Basic Call Functions: If selected, users can access menu options for call pickup,
park, unpark and transfer to mobile functions.
• Advanced Call Functions: If selected, users can access the menu options for do
not disturb, account code, withhold number and internal auto-answer functions. Note,
the Account Code menu is only shown if the system has been configured with
accounts codes.
• Forwarding: If selected, users can access the phone's menus for forwarding and
follow me functions.
• Hot Desk Functions: If selected, users can access the menu options for logging in
and out.
• Passcode Change: If selected, users can change their login code (security
credentials) through the phone menus..
• Phone Lock: If selected, users can access the menu options for locking the phone
and for setting it to automatically lock.
• Self Administration: If selected, users can access the phone’s Self-Administration
menu options.
• Voicemail Controls: If set, users can access the Visual Voice option through the
phone's Features menu.
SIP Phone Options
Application for Default = Equinox on Vantage
Vantage
Select the application to be used on Avaya Vantage™. The system supports Avaya
Vantage™ phones running either Avaya Vantage™ Connect or Avaya Workplace Client
applications as the dialer application. This field sets which application is indicated in
the auto-generated K1xxSupgrade.txt file the system provides to Avaya Vantage™
phones. If a mix of dialer applications is required, a static K1xxSupgrade.txt file
needs to be used. The options on the interface are:
• Equinox on Vantage: Select the option to use the Avaya Workplace Client client on
Avaya Vantage™ device.
• Vantage Basic/Connect: Select the option to use the Avaya Vantage™ Connect or
Avaya Vantage™ Basic applications on Avaya Vantage™ device.

Note:
This setting is not available for Avaya Vantage™ 3.0 version and above.

Related links
Telephony on page 239

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 Directory Services

Directory Services
Navigation: System | Directory Services
Related links
System on page 203
LDAP on page 259
HTTP on page 263

LDAP
Navigation: System | Directory Services | LDAP
Additional configuration information
For additional configuration information, see Centralized System Directory on page 612.
Configuration settings
LDAP (Lightweight Directory Access Protocol) is a software protocol for enabling anyone to locate
organizations, individuals, and other resources such as files and devices in a network. It can also
be used to import directory information.
The IP Office supports both LDAP V2 and LDAP V3:
• LDAP v2: This menu ( System > Directory Services > LDAP) supports LDAP v2 direct from
the IP Office service.
• LDAP v3: The Collaboration service on IP Office R11.1.2 and higher Linux-based IP Office
servers supports LDAP v3. For IP500 V2 servers, the Collaboration service is provided by
an IP Office Application Server. Using IP Office Web Manager, see Solution > Solution
Settings > User Synchronization Using LDAP.
Tip:
• IP Office systems also support the import of directory records from another IP Office
using HTTP. That includes using HTTP to import records that the other IP Office has
imported using LDAP.
LDAP records can contain several telephone numbers. Each will be treated as a separate
directory record when imported into the system directory.
An LDAP directory is organized in a simple "tree" hierarchy consisting of the following levels:
• The "root" directory (the starting place or the source of the tree), which branches out to
• Countries, each of which branches out to
• Organizations, which branch out to
• Organizational units (divisions, departments, and so forth), which branches out to (includes
an entry for)
• Individuals (which includes people, files, and shared resources such as printers)
An LDAP directory can be distributed among many servers. Each server can have a replicated
version of the total directory that is synchronized periodically. An LDAP server is called a Directory
System Agent (DSA). An LDAP server that receives a request from a user takes responsibility for

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System

the request, passing it to other DSA's as necessary, but ensuring a single coordinated response
for the user.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
LDAP Enabled Default = Off
This option turns LDAP support on or off. If the server being queried is an LDAP V3
server, support for LDAP V2 may need to be enabled on that server. LDAP V3 servers
typically support LDAP V2 but do not have it enabled by default.
User Name Default = Blank
Enter the user name to authenticate connection with the LDAP database. To determine
the domain-name of a particular Windows user look on the "Account" tab of the
user's properties under "Active Directory Users and Computers". Note that this means
that the user name required is not necessarily the same as the name of the Active
Directory record. There should be a built-in account in Active Directory for anonymous
Internet access, with prefix "IUSR_" and suffix server_name. Thus, for example, the
user name entered is this field might be: [email protected]
Password Default = Blank
Enter the password to be used to authenticate connection with the LDAP database.
Enter the password that has been configured under Active Directory for the above
user.
Alternatively, an Active Directory object may be made available for anonymous read
access. This is configured on the server as follows.
1. In Active Directory Users and Computers, enable Advanced Features under
the View menu.
2. Open the properties of the object to be published and select the Security tab.
3. Click Add and select ANONYMOUS LOGON and click Add and then OK
4. Click Advanced and select ANONYMOUS LOGON.
5. Click View/Edit and change Apply to to This object and all child objects.
6. Click OK to exit the menus.
7. Once this has been done on the server, any record can be made in the User
Name field in the System configuration form (however, this field cannot be
left blank) and the Password field left blank. Other non-Active Directory LDAP
servers may allow totally anonymous access, in which case neither User Name
nor Password need be configured.
Server IP Address Default = Blank
Enter the IP address of the server storing the database.
Server Port Default = 389
This setting is used to indicate the listening port on the LDAP server.
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 Directory Services

Field Description
Authentication Default = Simple
Method
Select the authentication method to be used. The options are:
• Simple: clear text authentication
• Kerberos: Not used.
Resync Interval Default = 3600 seconds. Range = 60 to 99999 seconds.
(secs)
The frequency at which the system should resynchronize the directory with the server.
This value also affects some aspects of the internal operation.
The LDAP search inquiry contains a field specifying a time limit for the search
operation and this is set to 1/16th of the resync interval. So by default a server should
terminate a search request if it has not completed within 225 seconds (3600/16).
The client end will terminate the LDAP operation if the TCP connection has been up
for more than 1/8th of the resync interval (default 450 seconds). This time is also
the interval at which a change in state of the "LDAP Enabled" configuration item is
checked.
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System

Field Description
Search Base Default = Blank
Search Filter These fields are used together to refine the extraction of directory records.
The Search Base specifies the point in the tree to start searching.
• The Search Base is a distinguished name in string form as defined in RFC1779.
The Search Filter specifies which objects under the base are of interest.
• The Search Filter deals with the attributes of the objects found under the Search
Base. It uses the format defined in RFC2254 except that extensible matching is not
supported.
• If left blank, the Search Filter defaults to (objectClass=*) which matches all
objects under the Search Base.
• You must ensure that the whole filter, and each object within the filter, are enclosed
within ( ) brackets.
The following are some examples applicable to an Active Directory database.
• To all user phone numbers in a domain:
- Search Base - cn=users,dc=acme,dc=com
- Search Filter - (telephonenumber=*)
• To restrict the search to a particular Organizational Unit (for example an office site)
and get cell phone numbers also:
- Search Base - ou=holmdel,DC=example,DC=com
- Search Filter - (|(telephonenumber=*)(mobile=*))
• To get the members of distribution list "group1":
- Search Base - cn=users,dc=example,dc=com
- Search Filter - (&(memberof=cn=group1,cn=users,dc=example,dc=com)
(telephonenumber=*))
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 Directory Services

Field Description
Number Attributes Default =
telephoneNumber,otherTelephone,homePhone=H,otherHomePhone=H,mobi
le=M,otherMobile=M
Enter the number attributes the server should return for each record that matches the
Search Base/Search Filter.
• Other Active Directory records are ipPhone, otherIpPhone,
facsimileTelephoneNumber, otherfacsimileTelephone Number, pager
or otherPager.
• The attribute names are not case sensitive.
• Other LDAP servers may use different attributes.
• The optional "=string" sub-fields define how that type of number is tagged in the
directory. Thus, for example, a cell phone number would appear in the directory as:
John Birbeck M 7325551234
Auto Populate MS Default = Enabled
Teams Data
When LDAP Enabled setting is enabled, the Auto Populate MS Teams Data setting
auto populates the Microsoft Teams URI obtained by IP Office in User | Mobility > MS
Teams URI and makes the MS Teams URI setting read-only.

Related links
Directory Services on page 259

HTTP
Navigation: System | Directory Services | HTTP
Additional configuration information
For additional configuration information, see Centralized System Directory on page 612.
Configuration settings
The system can use HTTP to import the directory records held by another system. Note that
support for HTTP can be disabled. The setting System | System | Avaya HTTP Clients Only can
restrict a system from responding to HTTP requests. The system's Unsecured Interfaces security
settings also included controls for HTTP access (HTTP Directory Read and HTTP Directory
Write).
For Server Edition, on Secondary Server, Expansion System (L) and Expansion System (V2)
systems, the HTTP settings are automatically defaulted to obtain the system directory from the
Primary Server.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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System

Field Description
Directory Type Default = None (No HTTP import)/IP Office SCN on Server Edition.
Set whether HTTP import should be used and the method of importation. The options
are:
• None: Do not use HTTP import.
• IP Office: Import from the system at the IP address set in the Source field.
• IP Office SCN: Import from a system in a multi-site network. The Source field is used
to select the Outgoing Line ID that matches the H.323 line to the remote system.
• Collaboration Services: When selected, other non configurable options are hidden or
their controls disabled with the enforced setting displayed.
Source Default = Blank/9999 on Server Edition.
The form of this field changes according to the Directory Type selection above. For IP
Office this field requires the IP address of the other system. For IP Office SCN, the
outgoing group ID of the IP Office line to the remote system is used.
List Default = All.
This field sets what types of directory record should be imported. The options are:
• All: Import the full set of directory records from the remote system.
• Config Only: Import just directory records that are part of the remote system's
configuration. Note that these will be treated as imported records and will not be added
to the local systems own configuration records.
• LDAP Only: Import just directory records that the remote system has obtained as the
result of its own LDAP import. This allows LDAP directory records to be relayed from
one system to another.
• HTTP Only: Import just directory records that the remote system has obtained as the
result of its own HTTP import. This allows HTTP directory records to be relayed from
one system to another.
URI Default = /system/dir/complete_dir_list?sdial=true
This field is for information only and cannot be adjusted. The path shown changes to
match the List setting above.
Resync Interval Default = 3600 seconds.
(secs)
Set how often the system should request an updated import. When a new import is
received, all previously imported records are discarded and the newly imported records
are processed.
HTTPS Enabled Default = On.
Turns HTTPS support on or off for directory record import.
Port Number Default = 443.
The port used for the Directory import.
When HTTPS Enabled is set to On, the default value is 443. When HTTPS Enabled is
set to Off, the default value is 80.

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 System Events

Related links
Directory Services on page 259

System Events
Navigation: System | System Events
The system supports a number of methods by which events occurring on the system can be
reported. These are in addition to the real-time and historical reports available through the System
Status Application (SSA).
Related links
System on page 203
Configuration on page 265
Alarms on page 266

Configuration
Navigation: System | System Events | Configuration
This form is used for general configuration related to system alarms.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
SNMP Agent Configuration
SNMP Enabled Default = Off.
Enables support for SNMP. This option is not required if using SMTP or Syslog.
Community Default = Blank.
(Read-only)
The SNMP community name to which the system belongs.
SNMP Port Default = 161. Range = 161, or 1024 to 65535. The port on which the system listens for
SNMP polling.
Device ID This is a text field used to add additional information to alarms. If an SSL VPN is
configured, Avaya recommends that the Device ID match an SSL VPN service Account
Name. Each SSL VPN service account name has an associated SSL VPN tunnel IP
address. Having the displayed Device ID match an SSL VPN service account name helps
identify a particular SSL VPN tunnel IP address to use for remotely managing IP Office.
Contact This is a text field used to add additional information to alarms.
Location This is a text field used to add additional information to alarms.

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Field Description
QoS Parameters
These parameters are used if the setting System | LAN1 | VoIP | Enable RTCP Monitor on Port 5005 is set to
On. They are used as alarm thresholds for the QoS data collected by the system for calls made by Avaya H.323
phones and for phones using VCM channels. If a monitored call exceeds any of the threshold an alarm is sent
to the System Status application. Quality of Service alarms can also be sent from the system using Alarms.
• The alarm occurs at the end of a call. If a call is held or parked and then retrieved, an alarm can occur for
each segment of the call that exceeded a threshold.
• Where a call is between two extensions on the system, it is possible that both extensions will generate an
alarm for the call.
• An alarm will not be triggered for the QoS parameters recorded during the first 5 seconds of a call.
Round Trip Delay Default = 350.
(msec)
Less than 160ms is high quality. Less than 350ms is good quality. Any higher delay will be
noticeable by those involved in the call. Note that, depending on the compression codec
being used, some delay stems from the signal processing and cannot be removed: G.711
= 40ms, G.723a = 160ms, G.729 = 80ms.
Jitter (msec) Default =20.
Jitter is a measure of the variance in the time for different voice packets in the same call to
reach the destination. Excessive jitter will become audible as echo.
Packet Loss (%) Default = 3.0.
Excessive packet loss will be audible as clipped words and may also cause call setup
delays.
Good Quality High Quality
Round Trip Delay < 350ms < 160ms
Jitter < 20ms < 20ms
Packet Loss < 3% < 1%

Related links
System Events on page 265

Alarms
Navigation: System | System Events | Alarms
These settings are not mergeable. Changes to these settings require a reboot of the system.
This form is used to configure what can cause alarms to be sent using the different alarm
methods.
• Up to 5 alarm traps can be configured for use with the SNMP settings on the System |
System Events | Configuration tab.
• Up to 3 email alarms can be configured for sending using the systems System | SMTP
settings. The email destination is set as part of the alarm configuration below.
• Up to 2 alarms can be configured for sending to a Syslog destination that is included in the
alarm settings.

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 System Events

Configuration Settings
Field Description
New Alarm This area is used to show and edit the alarm.
Destination
To use SNMP or Email the appropriate settings must be configured on the Configuration sub-tab. Note that the
Destination type is grayed out if the maximum number of configurable alarms destinations of that type has been
reached. Up to 5 alarm destinations can be configured for SNMP, 3 for SMTP email, and 2 for Syslog
Trap If selected, the details required in addition to the selected Events are:
• Server Address: Default = Blank. The IP address or fully qualified domain name
(FQDN) of the SNMP server to which trap information is sent.
• Port: Default = 162. Range = 0 to 65535. The SNMP transmit port.
• Community: Default = Blank The SNMP community for the transmitted traps. Must be
matched by the receiving SNMP server.
• Format: Default = IP Office. The options are:
- IP Office SNMP event alarms format in accordance with IP Office.
- SMGR SNMP event alarms format in accordance with SMGR.
Syslog If selected, the details required in addition to the selected Events are:
• IP Address: Default = Blank. The IP address of the Syslog server to which trap
information is sent.
• Port: Default = 514. Range = 0 to 65535. The Syslog destination port.
• Protocol: Default = UDP. Select UDP or TCP.
• Format: Default = Enterprise. The options are:
- Enterprise Syslog event alarms format in accordance with Enterprise.
- IP Office Syslog event alarms format in accordance with IP Office.
Email If selected, the details required in addition to the selected Events are:
Email: The destination email address.
Minimum Security Default = Warnings.
Level
The options are:
• Warnings: All events, from Warnings to Critical, are sent.
• Minor: Minor, major, and critical events are sent. Warnings are not sent.
• Major: Major and critical events are sent. Warnings and minor events will not be sent.
• Critical: Only critical events are sent.
Events Default = None
Sets which types of system events should be collected and sent. The table below lists the
alarms associated with each type of event. Text in italics in the messages is replaced with
the appropriate data. Items in [] brackets are included in the message if appropriate. The
subject line of SMTP email alarms takes the form "System name: IP address - System
Alarm".

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Type Events Event State Message

Entity Application Voicemail operation The Voicemail server is now operational.
Voicemail Failure The Voicemail server is down.
Voicemail Event - storage OK The Voicemail server storage is OK.
Voicemail Event - storage The Voicemail server storage is nearly full.
nearly full
Voicemail Event - storage full storage full The Voicemail server storage is
full.
Service Feature license missing Attempt to use a feature for which no license
is installed. License Type: <name>
All licenses in use The following licenses are all in use. License
Type: <name>
Clock source changed 8kHz clock source changed. Details will be
provided.
Logon failed Logon failure reason will be provided.
No free channels available No free channels were available. Outgoing
group ID: <number>
Hold music file failure Failed to load Hold Music source file.
All resources in use The following system resources are all in
use: <resource type> will be provided.
OEM card slot error System running secondary software or error
description with OEM card will be provided.
Network interconnect failure Details of the network interconnection failure
will be provided.
SIP message too large SIP message Rx error - too large - ignored.
Contact Change The PC card in name has changed.
Flash Card
Expansion Operational Expansion module name link is up.
Module Failure Expansion module name link is down.
Error Expansion module name link has a link error.
Change Expansion module name link has changed.
Trunk Operational Trunk number (name) [on expansion module
number] is now operational.
Failure Trunk number (name) [on expansion module
number] is down.
Trunk Trunk seize failure Seize failure: Channel [number] or Port
[number].
Incoming call outgoing trunk Incoming call outgoing trunk: Channel
failure [number] or Port [number].
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 System Events

Type Events Event State Message

CLI not delivered CLI not delivered: Channel [number] or Port
[number].
DDI incomplete DDI incomplete. Expected Number of
digits: .
LOS LOS
OOS OOS
Red Alarm Red Alarm
Blue Alarm Blue Alarm
Yellow Alarm Yellow Alarm
IP connection failure IP connection failure. IP Trunk Line Number:
<number> or Remote end IP address: <IP
address>
Small Community Network Small Community Network invalid
invalid connection connection. IP trunk line number: <number>
or remote end IP address: <IP address>
Link Device changed Device changed. Home Extension Number: .
LDAP server communication LDAP server communication failure
failure
Resource down Link/resource down. Module type, number
and name will be provided.
SMTP server communication SMTP server communication failure
failure
Voicemail Pro connection Voicemail Pro connection failure
failure
Dialer connection failure The Dialer connection has been lost.
VCM Operational VCM module name is now operational.
Failure VCM module name has failed.
Memory Card Invalid Card
Free Capacity
Generic Generic Non-primary location boot System running backup software.
alarm
Invalid SD Card Incompatible or Invalid (System or Optional)
SD Card fitted.
Network link failure Network Interface name (ip address) has
been disconnected.
Network link operational Network Interface name (ip address) has
been connected.
System warm start System has been restarted (warm start).
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Type Events Event State Message

System cold start System has restarted from power fail (cold
start).
SNMP Invalid community Invalid community specified in SNMP
request.
License License Server Operational The license server is now operational.
Server
Server failure The license server is no longer operational.
License Key License Key Failure
Failure
Loopback Loopback Near end line loopback Trunk number (name) [on expansion module
number] is in near end loopback.
Near end payload loopback Trunk number (name) [on expansion module
number] is in near end loopback with
payload.
Loopback off Trunk number (name) [on expansion module
number] has no loopback.
Phone Phone Phone has been unplugged The phone with id n has been removed from
Change Change extension extension (unit, port number).
Phone has been plugged in The phone with type type (id number) has
been plugged in for extension extension
(unit, port number).
Quality of QoS If Enable RTCP Monitor on Port 5005 is selected, any monitored calls that
Service Monitoring exceeds the set QoS Parameters causes an alarm.
Syslog Basic Audit Events as written to the system Audit Trail. Available on Syslog output only.
Extended Configuration change information. Each message contains one configuration
Audit or security settings object attribute change, and optionally the previous and
new values.
System If selected, System monitor traces are packed into Syslog traces.
Monitor
System Configuration Small Community Network dial Small Community Network dial plan conflict
plan conflict
No incoming call route for call The following line had no Incoming Call
Route for a call. Line: <number> or Line
Group ID: <number>.
Installed hardware failure Installed hardware failure details will be
provided.
System
Shutdown
Running
Backup
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 SMTP

Type Events Event State Message

Emergency Emergency call successful Successful Emergency Call | Emergency
Calls call! Location:location Dialled:dialled number
Called:number sent on the line CallerID: ID
Usr:user Extn:extension
Emergency call failed Failed Emergency Call | Emergency
call! Location:location Dialled:dialled
numberFailCause:cause Usr:user Extn:
extension

Alarm Types
Note the following.
• Voicemail Pro Storage Alarms: The alarm threshold is adjustable through the Voicemail Pro
client.
• Embedded Voicemail Storage Alarms: A disk full alarm is generated when the Embedded
Voicemail memory card reaches 90% full. In addition a critical space alarm is generated at
99% full and an OK alarm is generated when the disk space returns to below 90% full.
• Loopback: This type of alarm is only available for systems with a United States locale.
The list of IP Office alarms is available on the Admin CD in the folder \snmp_mibs\IPOffice.
Related links
System Events on page 265

SMTP
Navigation: System | SMTP
These settings are not mergeable. Changes to these settings require a reboot of the system.
Configuration Settings
SMTP can be used as the method of sending system alarms. The email destination is set as part
of the email alarms configured in System | System Events | Alarms.
SMTP can be used with Embedded Voicemail for Voicemail Email. The voicemail destination is set
by the user's Voicemail Email address.
Field Description
Server Address Default = Blank
This field sets the IP address of the SMTP server being used to forward SNMP
alarms sent by email.
Port Default = 25. Range = 0 to 65534.
This field set the destination port on the SMTP server.
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Field Description
Email From Address Default = Blank
This field set the sender email address. Depending of the requirements of the
SMTP server this may need to be a valid email address hosted by that server.
Otherwise the SMTP email server may need to be configured to support SMTP
relay.
Use STARTTLS Default = Off. (Release 9.0.3).
Select this field to enable TLS/SSL encryption. Encryption allows voicemail-to-
email integration with hosted email providers that use secure transport.
Server Requires Default = Off
Authentication
Select if the SMTP server requires authentication. When selected, the following
fields become available
User Name Default = Blank
Sets the user name for SMTP server authentication.
Password Default = Blank
Sets the password for SMTP server authentication.
Use Challenge Response Default = Off.
Authentication (CRAM-
Selected if the SMTP server uses CRAM-MD5.
MD5)

Related links
System on page 203

SMDR
Navigation: System | SMDR
The system can be configured to output SMDR (Station Message Detail Reporting) records for
each completed call.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 SMDR

Field Description
Output Default = No Output.
Select the type of call record that the system should create. The options are:
• No Output – Do not generate SMDR records.
• SMDR Only – Generate SMDR records and send those records using the settings
below.
• Hosted Only - Used for subscriptions systems only. Stores the system's SMDR
records on the cloud services supporting the system. Specific users can be configured
to access those settings through the user portal.
SMDR: Station Message Detail Recorder Communications
This fields are available when SMDR is selected as the output. For information on SMDR record details, see the
SMDR appendix.
IP Address Default = 0.0.0.0 (Listen).
The destination IP address for SMDR records. Each time a new record is generated, the
system will attempt to send the record to the address specified.
• The address 0.0.0.0 puts the system into listen mode. Using an application such as
HyperTerminal or Putty, a TCP/IP connection to the system’s IP address and specified
TCP port will collect any new and or buffered records.
• Any other address puts the system into send mode. Each time a new record is
generated, the system attempts to send the record to the specified address and port
using a TCP/IP connection. If the connection is not successful, the record is buffered
(see below) until a successful connection occurs for a subsequent new record.
TCP Port Default = 0.
The IP port for sending or collecting SMDR records.
Records to Buffer Default = 500. Range = 10 to 3000.
The system buffers new records when there is not TCP/IP connection. It can buffer up to
3000 SMDR records.
If the cache is full, the system discards the oldest record each time a new record is
added.
Call Splitting for Default = Off.
Diverts
When enabled, for calls forwarded off-switch using an external trunk, the SMDR
produces separate initial call and forwarded call records:
• The two sets of records have the same Call ID.
• The Call Start Time fields of the forwarded call records are reset from the moment of
forwarding on the external trunk.
This applies for:
• Calls forwarded by forward unconditional, forward on no answer, forward on busy, DND
or mobile twinning.
• Calls forwarded off-switch by an incoming call route.

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Related links
System on page 203

VCM
Navigation: System | VCM
This form allows adjustment of the operation of any Voice Compression Modules (VCM's) installed
in a control unit.
Calls to and from IP devices can require conversion to the audio codec format being used by the
IP device. For systems this conversion is done by voice compression channels. These support the
common IP audio codecs G.711, G.723 and G.729a. For details of how to add voice compression
resources to a system, refer to the IP Office Installation Manual.
These settings should only be adjusted under the guidance of Avaya support.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
When are Voice Compression Channels Used
IP Device to Non-IP Device: These calls require a voice compression channel for the duration of
the call. If no channel is available, busy indication is returned to the caller.
IP Device to IP Device: Call progress tones (for example dial tone, secondary dial tone, etc) do
not require voice compression channels with the following exceptions:
• Short code confirmation, ARS camp on and account code entry tones require a voice
compression channel.
• Devices using G.723 require a voice compression channel for all tones except call waiting.
When a call is connected:
• If the IP devices use the same audio codec no voice compression channel is used.
• If the devices use differing audio codecs, a voice compression channel is required for each.
Non-IP Device to Non-IP Device: No voice compression channels are required.
Music on Hold: This is provided from the system's TDM bus and therefore requires a voice
compression channel when played to an IP device.
Conference Resources and IP Devices: Conferencing resources are managed by the
conference chip which is on the system's TDM bus. Therefore, a voice compression channel
is required for each IP device involved in a conference. This includes services that use conference
resources such as call listen, intrusion and silent monitoring. They also apply to call recording.
Page Calls to IP Device: Page calls require 1 voice compression channel per audio codec being
used by any IP devices involved. The system only uses G.729a for page calls, therefore only
requiring one channel but also only supporting pages to G.729a capable devices.
Voicemail Services and IP Devices: Calls to the system voicemail servers are treated as
data calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice
compression channel.

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 VCM

Fax Calls: These are voice calls but with a slightly wider frequency range than spoken voice
calls. The system only supports fax across IP between systems with the Fax Transport option
selected.
SIP Calls:
• SIP Line Call to/from Non-IP Devices: Voice compression channel required.
• Outgoing SIP Line Call from IP Device: No voice compression channel required.
• Incoming SIP Line Call to IP Device: Voice compression channel reserved until call
connected.
T38 Fax Calls: The system supports T38 fax on SIP trunks and SIP extensions. Each T38 fax
call uses a VCM channel.
• Within a multi-site network, an T38 fax call can be converted to a call across across an H.323
line between systems using the Fax Transport Support protocol. This conversion uses 2
VCM channels.
• In order use T38 Fax connection, the Equipment Classification of an analog extension
connected to a fax machine can be set Fax Machine. Additionally, the short code feature
Dial Fax is available.
Measuring Channel Usability
The System Status Application can be used to display voice compression channel usage. Within
the Resources section it displays the number of channel in use. It also displays how often there
have been insufficient channels available and the last time such an event occurred.
Field Description
Echo Return Loss (dB) Default = 6dB. IP500 VCM, IP500 VCM V2 and IP500 Combination Cards. This
control allows adjustment of expected echo loss that should be used for the echo
cancellation process.
Echoes are typically generated by impedance mismatches when a signal is
converted from one circuit type to another, most notably from analog to IP. To
resolve this issue, an estimated echo signal can be created from one output and
then subtracted from the input to hopefully remove any echo of the output.
The options are: 0dB, 3dB, 6dB and 9dB.
Nonlinear Processor Default = Adaptive. I
Mode
A low level of comfort noise is required on digital lines during periods where there
would normally be just silence. This is necessary to reassure users that the call is
still connected. These controls allow adjustment of the comfort noise generated by
the nonlinear processor (NLP) component of the VCM. The options are:
• Adaptive: Adaptive means the comfort noise generated by the NLP will try to
match background noise.
• Silence: Silence means the NLP will not generate comfort noise at all
• Disabled: Nonlinear processing is not applied, in which case some residual echo
may be heard.
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System

Field Description
NLP Comfort Noise Default = -9dB.
Attenuation
The options are: -3dB, -6dB and -9dB.
NLP Comfort Noise Default =-30dB.
Ceiling
The options are: -30dB and -55dB.
Modem
For Fax relay, these settings allow adjustment of the TDM side operation applied to fax calls using VCM
channels.
Tx Level (dB) Default = -9dB. Range = 0 to -13dB.
CD Threshold Default = -43dB, Options = -26dB, -31dB or -43dB.
No Activity Timeout Default = 30 seconds. Range = 10 to 600 seconds.
(secs)

Related links
System on page 203

Integrated Contact Reporter

Navigation: System | Contact Center | Integrated Contact Reporter
Integrated Contact Reporter is a simple Contact Center application that involves five to twenty
five agents and up to five supervisors. The application collects and displays information on the
current status of hunt groups and users that have been configured for Integrated Contact Reporter
operation.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Field Description
Busy Not Available Reason Codes
Agents who indicate that they are in a 'busy not available' state can be prompt to also indicate the reason for
being in that state. This menu allows descriptions for the possible reasons to be entered. The descriptions are
then used in menus from which the Agent's make selections when setting themselves into busy not available
state and in reports on Agent status.
Code/Reason Rows 1 to 8 can be used to contain descriptions of up to 31 characters each. Rows
0 and 9 are fixed as Unsupported and Busy Not Available.
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 VoIP

Field Description
Default After Call Work Default = 10. Range = 10 to 999 seconds.
Time (seconds)
If an agent goes into the After Call Work (ACW) state, either automatically or
manually, this field sets the duration of that state after which it is automatically
cleared. This duration can be overridden by the Agent's own setting (User |
Telephony | Supervisor Settings | After Call Work Time). During ACW state, hunt
group calls are not presented to the user.

Related links
System on page 203

VoIP
Navigation: System | VoIP
These menus apply to the IP Office system's VoIP operation.
Related links
System on page 203
VoIP on page 277
VoIP Security on page 279
Access Control Lists on page 282

VoIP
Navigation: System | System | VoIP | VoIP
This tab is used to set the codecs available for use with all IP (H.323 and SIP) lines and
extensions and the default order of codec preference.
• Avaya H.323 telephones do not support G.723 and will ignore it if selected.
• For systems with H.323 lines and extensions, one of the G.711 codecs must be selected and
used.
• G.723 and G.729b are not supported by Linux based systems.
• The number of channels provided by an IP500 VCM 32 or IP500 VCM 64 card, up to a
maximum of 32 or 64 respectively, depends on the actual codecs being used. This also
applies to IP500 VCM 32 V2 and IP500 VCM 64 V2 cards. The following table assumes that
all calls using the VCM use the same codec.
Codec IP500 VCM 32 IP500 VCM 32 V2 IP500 VCM 64 IP500 VCM 64 V2
G.711 32 64
G.729a 30 60
G.723 22 44
G.722 30 60

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System

Paging from an IP device uses the preferred codec of that device. It is the system administrator's
responsibility to ensure all the target phones in the paging group support that codec.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
Ignore DTMF Mismatch Default = Enabled.
for Phones
When enabled, the following settings are visible and configurable:
• Extension | H.323 Extension | VoIP | Requires DTMF
• Extension | SIP Extension | VoIP | Requires DTMF
When enabled, during media checks, the system ignores DTMF checks if the call is
between two VoIP phones and the extension setting Requires DTMF is set to Off.
The two phones can be located on different systems in a Server Edition or SCN
deployment.

Note:
Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
Allow Direct Media Default = Off.
Within NAT Location
When enabled, the system allows direct media between devices that reside behind
the same NAT. Devices are behind the same NAT if their public IP addresses are the
same.

Note:
Direct media is not be possible if other settings, such as codecs, NAT settings,
or security settings, are mismatched.
The default behavior is to allow direct media between all types of devices (H323
and SIP remote workers and IP Office Lines behind a NAT). In the case of
routers that have H323 or SIP ALG, it can be desirable to allow direct media only
between certain categories of devices. This can be configured by adding the NoUser
Source Number MEDIA_NAT_DM_INTERNAL. For information, see User | Source
Numbers.
Disable Direct Media Default = cleared
For Simultaneous
The user logged into the IP softphone client uses virtual extension records. The
Clients
Disable Direct Media For Simultaneous Clients setting is used to set the default
Allow Direct Media Within NAT Location setting behavior of the virtual extensions.
When Disable Direct Media For Simultaneous Clients setting is enabled, the
system disables direct media for all the clients logged in simultaneously.

Note:
Enabling the Disable Direct Media For Simultaneous Clients settings
disables the Allow Direct Media Within NAT Location settings for virtual
extension records used by IP softphones.
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 VoIP

Field Description
RFC2833 Default Default = 101. Range = 96 - 127.
Payload
This field specifies the default value for RFC2833 dynamic payload negotiation.
Service providers that do not support dynamic payload negotiation may require a
fixed value.
OPUS Default Payload Default = 116.
This field specifies the default value and the range to be used for Opus codec.
This field is only used for Linux-based systems.

Note:
This field is not avialable on IP500v2, but the Unknown Codec passthrough
and the OPUS settings are available to set individually.
Available Codecs This list shows the codecs supported by the system and those selected as usable.
Those codecs selected in this list are then available for use in other codec lists
shown in the configuration settings. For example, the adjacent Default Selection list
and the individual custom selection list on IP lines and extensions.

Warning:
Removing a codec from this list automatically removes it from the codec lists of
any individual lines and extensions that are using it.
The supported codecs (in default preference order) are: Opus, G.711 A-Law, G.711
U-Law, G.722, G.729, and G.723.1. The default order for G.711 codecs varies to
match the default companding settings of the system. G.723.1 and G.729b are not
supported on Linux- based systems.
Default Codec By default, all IP (H.323 and SIP) lines and extensions added to the system have
Selection their Codec Selection setting set to System Default. That setting matches the
codec selections made in this list. The buttons between the two lists can be used to
move codecs between the Unused and the Selected parts of the list and to change
the order of the codecs in the selected codecs list.

Related links
VoIP on page 277

VoIP Security
Navigation: System | System | VoIP | VoIP Security
Use to set system level media security settings. These settings apply to all lines and extensions
on which SRTP is supported and which have their Media Security settings configured to be Same
as System. Individual lines and extensions have media security settings that can override system
level settings.
Simultaneous SIP extensions that do not have physical extensions in the configuration use the
system security settings.
SM lines and all centralized user extensions must have uniform media security settings.
These settings are not mergeable. Changes to these settings require a reboot of the system.

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System

Name Description
Default Extension Default = Extension password set during initial configuration.
Password
This default extension password is automatically assigned to each H.323 and SIP
extension entry when they are added to the system configuration. Each extension's
password can be changed through the extension's own settings if required.
The extension password is used for registration of IP phones with the system. The
password must be 9 to 13 digits. Use the 'eye' icon to see the existing default password.
Media Security Default = Disabled.
Secure RTP (SRTP) can be used between IP devices to add additional security. These
settings control whether SRTP is used for this system and the settings used for the
SRTP. The options are:
• Disabled: Media security is not required. All media sessions (audio, video, and data) is
enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
If media security is enabled (Enforced or Preferred), we recommend that you enable a
matching level of security using System | LAN | VoIP | H.323 Signalling over TLS.
The endpoints that support Secure RTP are:
• IP Office , SIP and SM lines
• Avaya H.323 extensions: 9608, 9611, 9621, 9641
• Avaya SIP extensions: 9608, 9611, 9621 and 9641 (in centralized branch
deployments), 1100 Series, 1200 Series, B179, E129, H175, J100 Series, K100 Series
(Vantage), Scopia XT series
• 3rd Party SIP extensions that support SRTP
Media Security Not displayed if Media Security is set to Disabled. The options are:
Options
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be protected
using encryption. The default is to encrypt just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be protected
using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
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 VoIP

Name Description
Strict SIPS Default = Off.
This setting is available in Enterprise Branch deployments only. This option provides a
system-wide configuration for call restrictions based on SIPS URI.
When this option is off, calls are not rejected due to SIPS. A call is sent according to the
configuration of the outgoing trunk or line that it is routed to, regardless of the way the
call came in, even if the call came in as a SIP invite with SIPS URI and is being sent with
a SIP URI onto a non-secure SIP trunk.
When this option is on, an incoming SIP invite with SIPS URI if targeted to a SIP trunk
(SM line or SIP line) is rejected if the target trunk is not configured with SIPS in the URI
Type field.

Note:
• Strict SIPS is not supported with 9600 Series and J100 Series SIP Feature
phones.

Calling Number Verification

These settings configure the SIP trunks use of STIR protocols for calling number verification.
For more details, see SIP Calling Number Verification (STIR/SHAKEN) on page 866.
Field Description
Incoming Calls Default = Allow Not Failed
Handling
Sets the defaults for which calls are accepted by the system based on the authentication
level of the call. This default can be overridden in the individual line configuration.
• Allow All - Allow all calls regardless of calling number verification.
• Allow Validated - Only accept verified calls with full or partial attestation.
• Allow Not Failed - Accept all calls expect those that specifically failed verification.
Note this can include calls with no reported verification result.
Validation Default = Off
Presentation
If enabled, the system will prefix the caller ID information displayed on phones with a
character indicating the result of the call's validation result. This will be:
• A tick mark for full verification.
• A question mark for partial verification.
• A cross for authentication failed.
When enabled, the system will also inspect the display information on all received trunk
calls to ensure they do not start with these characters in order to avoid spoofing.

Related links
VoIP on page 277

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System

Access Control Lists

Navigation: System | System | VoIP | Access Control Lists
Name Description
SIP UA Blacklist The list sets SIP User Agent (UA) strings that are blocked when the relevant LAN's
System > LANx > VoIP > Allowed SIP User Agents setting is set to Block
Blacklist only.
• Not supported on IP500 V2 systems.
SIP UA Whitelist This lists sets the SIP User Agent (UA) strings allowed to register when the relevant
LAN's System > LANx > VoIP > Allowed SIP User Agents setting is set to Avaya
Clients & Whitelisted or Whitelisted Only.
• Not supported on IP500 V2 systems.
IP Whitelist The system can automatically block traffic from an IP address based on too many
failed registration attempts from that address. This list can be used to create a list of
addresses which should not be blocked.
• This can be useful when there are multiple devices registering from behind the
same single public IP address. In such a scenario, there can a higher incidence of
failed registrations.
• Supported on IP500 V2 systems for R11.1 FP2 and higher.

Related links
VoIP on page 277

Dialer
Navigation: System | Dialer
Use to configure the functions required for an Outbound Contact Express deployment.
These settings are mergeable. However, changes to the Operation field or to the Trunk Range /
IP Office table require a reboot.
It is recommended that you do not change the mergeable settings while the system is in use.
Field Description
Operation Default = Off.
On the primary IP Office Server Edition server, set this field to Primary. For all other
IP Office servers, set this field to Child. When set to Off or Child, no other fields are
displayed.
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 Dialer

Field Description
Record Mode Default = Off
Defines the automatic call recording function on VMPro. The options are:
• Whole Call: The entire call is recorded.
• Agent Connected: Recording starts once the conversation begins.
• Off
Record Controls Default = Full
Defines what functions an agent can perform from WebAgent or from the handset. The
options are:Full, Pause or Off.
Record Mode and Record Mode and Record Controls are related. The combined configuration settings
Record Controls are listed below.
Note that stopping and starting the recording creates multiple recording files. Pausing
and resuming the recording keeps the recording in a single file.
Record Record Result
Mode Controls
Off Off Calls are not recorded.
Agent Off All calls are always recorded from the time the agent joins the call.
Connect
ed
Agent Pause All calls are always recorded but the Agent can pause and resume
Connect recording.
ed
Agent Full All calls are always recorded from the time the Agent joins the
Connect call. Agent has full control on when calls get recorded.
ed
Whole Off All calls are always recorded from the time the customer answers.
Call
Whole Pause All calls are always recorded from the time the customer answers
Call but the Agent can pause and resume the recording.
Whole Full Call recording starts before the agent is connected. All calls
Call are always recorded but the Agent can pause and resume the
recording
Agent Call Back Default = 60. Range = 30 - 300.
Time
The number of seconds an agent has to make a manual call after a customer hang up.
Used when a customer wants to be called on a different number.
Remote Agent Default = Blank. Maximum length = 33.
Display Text
Specify the text string displayed on the remote agent extension if that extension supports
displays and the protocol allows it to be transmitted.
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System

Field Description
Remote Agent Default = Blank. Maximum length = 31.
Confirmation Voice
Specify the Call Flow Entry point name used to play a greeting to the remote agent when
Prompt
they log in. The actual Entry Point is added as a Modules Entry point using the VMPro
Client. The entry point cannot be added as a short code, user or group entry point.
Remote Agent First Default = 0. The first extension number allocated to a remote agent. It must not conflict
Extension Number with the existing dialing plan. If the range contains existing user extensions, they are
used when assigning extensions to remote users.
Remote Agent Default = 0. Maximum = 500.
Number of
The range of extensions starting from the one above. A user is created for every
Extensions
extension. If the field is edited and the number of extensions is reduced, the number
of remote agents that can log in is reduced to the new setting. However, reducing the
range does not automatically delete previously created users. Users can only be deleted
manually.
Use Custom Hold Default = unchecked. Defines system behavior when a call is placed on hold. When
Treatment unchecked the the system Hold Music setting is used for the system's music on hold
source. When checked, the music on hold source is VMPro.
Record while on Default = unchecked. When the Use Custom Hold Treatment box is checked, the
Hold Record while on Hold setting can be enabled. When unchecked, recording is paused
when the call is on hold. When checked, recording continues when the call is on hold.
Trunk Range / IP The number of trunks used by Outbound Contact Express. The default entry is Trunk
Office Range: 1-250 for the Primary (Local) server. 250 is the maximum number of trunks
configured on a single server. Use this table to define the number of trunks managed by
the Primary and Secondary systems. The trunk range must match the line numbers used
by the Proactive Contact Dialer. Enter only one range per server.

Related links
System on page 203

Contact Center
Navigation: System | Contact Center
The Contact Center tab contains the user information required by IP Office to synchronize account
information with an Avaya Contact Center Select (ACCS) system. The information is synchronized
using the Contact Center Management Application (CCMA). These settings are only used for the
deployment of an ACCS system.
This tab is visible on the Server Edition Primary Server and Standard Mode IP500 V2 systems.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Contact Center

Field Description
Contact Center Default = None.
Application
The options are:
• Avaya Contact Center Select
• Avaya IP Office Contact Center
• Integrated Contact Reporter (not supported in IP Office Release 11.0)
Synchronize to Default = Off.
this System
When set to On, the CCMA fields below are enabled.
CCMA Address Default = Blank
Address of the Contact Center Management Application system.
CCMA Username Default = Blank
User name on the Contact Center Management Application system.
CCMA Password Default = Blank
Password on the Contact Center Management Application system.
Default After Call Applicable for Integrated Contact Reporter
Work Time
Default = 10 seconds, Minimum = 10 seconds, Maximum = 999 seconds
The default time set for After Call Work (ACW). If configured, ACW begins at the end of a
call. Hunt group calls are not sent to the agent during ACW.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Busy Not Applicable for Integrated Contact Reporter
Available Reason
Default = 2 codes
Codes
Maximum = 9 codes
The reasons for 0 and 9 are assigned by default and cannot be modified. You can
configure the rest.
Use the configure icon to add Busy Not Available reasons and assign them to the
available codes.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Related links
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System

Remote Operations
Navigation: System > Remote Operations
In addition to monitoring the status and alarms of a subscription mode IP Office system, Customer
Operations Management (COM) can support a number of additional services for the IP Office
system. For details, refer to Using Customer Operations Manager for IP Office Subscription
Systems.
Settings Description
Remote Access This option supports HTTPS, SFTP, SSH and RDP connections to IP Office servers
managed by Customer Operations Management.
Co-located Servers This option allows Remote Access support to extend to other servers on the same
network as the IP Office system. That includes connection to standalone IP Office
Application servers.
This option requires configuration of a TCP tunnel for each connection through the
System > Services > Remote Support Services menu.
Remote Upgrade/ This option supports backup and restoration from IP Office to COM. Enabling the
Backup Remote Upgrade/Backup setting allows automatic daily backups.
Centralized This option supports remote connections to IP Office servers using IP Office admin tools
Management (System Status Application, SysMonitor and IP Office Web Manager).
Centralized This option supports the uploading and storage of system log files to COM.
Diagnostics Log

Related links
System on page 203

Avaya Cloud Services

Navigation: System > Avaya Cloud Services
The Avaya Cloud Services tab contains configuration settings for Avaya Cloud Services and
features that use Avaya Cloud Services. For details, see the IP Office Avaya Workplace Client
Installation Notes manual.
• Features that use Avaya Spaces are not supported for Avaya Workplace Client when using
IPv6.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Avaya Cloud Services

Field Description
Profile Name Default = None
This name is used to identify the IP Office in the profile settings written into Avaya Cloud
services if Enable Settings File URL Sync is enabled.
• Within a multi-site network, the name must be unique.
Enable Avaya Default = Disabled
Cloud Account
Enable interoperation between the IP Office and Avaya Cloud Services.
• You must also add the Avaya Spaces API Key and Avaya Spaces Key Secret for the
customer domain to the IP Office security settings.

AVAYA CLOUD ACCOUNT CONFIGURATION

Field Description
Account URL Ensure that the URL matches the appropriate value below:
• IP500 V2 = accounts-ipo.avayacloud.com
• Linux-based Server = accounts.avayacloud.com
Company Domain Default = Blank
The company domain registered and verified with Avaya Spaces.

USER SYNCHRONIZATION
Field Description
Enable user sync Default = Disabled
If enabled, the IP Office system automatically synchronizes user information with Avaya
Spaces.
Manual user sync Default = Disabled
This option is only available in IP Office Web Manager.
• You can use the Refresh button to request an manual synchronization.
• The Synchronization Status field shows the result of the last synchronization.

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System

SETTINGS FILE URL SYNCHRONIZATION

Field Description
Enable Settings Default = Disabled
File URL Sync
Controls whether the IP Office sends its SIP FQDN and Profile Name to Avaya Cloud
Services.
• Avaya Cloud Services uses the information to write a profile for the IP Office containing
its profile name and 46xxsettings.txt file address. The allows Avaya Workplace
Client users to connect to the IP Office using their Unique Identity email address.
• Avaya Cloud Services needs address details for each IP Office that is acting as a SIP
registrar.
The possible settings are:
• Enable for IP Office current Node
Send settings file information for just the current IP Office system.
• Enable for all IP Office Nodes
Send settings file information for all IP Office systems in the network.
• Disabled
Do not send settings file information from the IP Office.

AVAYA CLOUD AUTHORIZATION

Cloud authorization enables users to login to Avaya Workplace Client using a single-sign on
(SSO) account such as their Google, Office 365, or Salesforce account. It also allows initial Avaya
Workplace Client registration using the user' email address.
• When using IP Office Web Manager to manage a multi-site network (not SCN), you can
use the Solution > Actions > Synchronize Single Sign-On configuration command to
synchronize these settings on other servers with those on the primary server.
Field Description
Enable Avaya Default = Disabled
Cloud Account
Control whether cloud authorization is enabled.
Authorization
• Avaya Cloud Account Authorization requires TLS between the IP Office and the Avaya
Workplace Client.
Token Cache Default = 15 minutes. Range = 15 to 60 minutes.
Time
The time in minutes that the IP Office caches authorization tokens from Avaya Cloud
Services.

Related links
System on page 203

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 Avaya Push Notification Services

Avaya Push Notification Services

Navigation: System > Avaya Push Notification Services
Push notification is used to send Avaya Workplace Client users on Apple iOS devices notification
of new calls and voicemail messages. Push notifications also requires Avaya Cloud Services on
page 286 to be enabled.
For full details, refer to the IP Office Avaya Workplace Client Installation Notes manual.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Enable Apple Default = Off
Push Notification
If enabled, the IP Office system will use push notifications for iOS Avaya Workplace Client
users.
• Avaya Cloud Account Authorization requires TLS between the IP Office and the Avaya
Workplace Client.
• When using IP Office Web Manager to manage a multi-site network (not SCN), you can
use the following commands to synchronize settings on other servers with the primary
server:
- Solution > Actions > Synchronize APNS configuration synchronizes the Enable
Apple Push Notification setting.
- Solution > Actions > Synchronize APNP System-ID synchronizes the System-ID,
Avaya Spaces API Key and Avaya Spaces Key Secret settings.
• The System-ID is a hidden value generated by an IP Office when the Enable Apple
Push Notification setting is enabled.
Avaya Push Default = pnp.avaya.com
Notification
This setting is for information only, not editable. This is the address of the Avaya service
Provider Address
to which the IP Office sends push notifications. The service forwards those notifications to
the Apple Push Notification service which forwards notifications the iOS devices.
Payload Default = On
Encryption
This setting is for information only, not editable.
Push Notification Default = com.avaya.AvayaCommunicator
Application Type
This setting is for information only, not editable. This is the application string
for which push notifications are sent. Avaya Workplace Client still uses the older
com.avaya.AvayaCommunicator string.

Related links
System on page 203

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Chapter 26: Line

The line settings shown in the system configuration will change according to the types of trunk cards
installed in the control unit or added using external expansion modules.
Warning:
Changing Trunk Cards - Changing the trunk card installed in a control unit will result in line
settings for both the previous trunk card and the installed trunk card. To change the type of trunk
card installed in a particular card slot, the configuration must be defaulted. This does not apply
if replacing an existing card with one of a higher capacity or fitting a trunk card into an unused
slot.
Trunk Incoming Call Routing
Trunks are categorized as external or trunk. The trunk type affects how the system routes calls
received on that trunk and the routing of calls to the trunk.
Trunk Types Incoming Calls Routed by
External Trunks • Incoming calls are routed by matching call details
against the settings of the system Incoming Call
• Analog trunks
Routes.
• T1 Robbed Bit
• Line short codes are not used.
• E1R2
• ISDN BRI (excluding So)
• ISDN PRI T1
• ISDN PRI E1
• SIP
Internal Trunks Incoming calls are routed by looking for a match to the
incoming digits in the following order:
• QSIG (T1, E1 or H.323)
• Extension number.
• BRI So
• Trunk short codes (excluding ? short code).
• H.323
• System short codes (excluding ? short code).
• SCN
• Trunk ? short code.
• SM
• System ? short code.
• IP Office Line

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 Line Groups
Each system trunk (or in some cases individual trunk channels) can be configured with an Incoming
Group ID and an Outgoing Group ID. These group IDs are used as follows:
• Incoming Call Routes - For incoming calls on external trunks, the Incoming Group ID of the
trunk is one of the factors used to match the call to one of the configured incoming call routes.
• Short Codes - For dialing which matches a short code set to a Dial feature, the short codes
Line Group ID can indicate either an ARS form or to use a trunk from set to the same
Outgoing Group ID. If the call is routed to an ARS form, the short codes in the ARS form will
specify the trunks to use by matching Outgoing Group ID.
Removing Unused Trunks
In cases where a trunk card is installed but the trunk is not physically connected, it is important to
ensure that the trunk is disabled in the configuration. This can be done on most trunks using by
setting the line's Admin setting to Out of Service.
This is especially important with analog trunks. Failure to do this may cause the system to attempt to
present outgoing calls to that trunk. Similarly, where the number of channels subscribed is less than
those supportable by the trunk type, the unsubscribed channels should be disabled.
Clock Quality
Calls between systems using digital trunks (for example E1, E1R2, T1 PRI and BRI) require an
common clock signal. The system will try to obtain this clock signal from a PSTN exchange through
one of its digital trunks. This is done by setting the Clock Quality setting of that line to Network.
If there are multiple trunks to public exchanges, another trunk can be set as Fallback should the
primary clock signal fail. Other trunks should be set as Unsuitable.
Related links
ACO Line on page 292
Analog Line on page 297
BRI Line on page 305
H.323 Line on page 311
IP DECT Line on page 317
IP Office Line on page 322
Legacy SIP DECT Line on page 331
MS Teams Line on page 334
PRI Trunks on page 343
SIP Line on page 372
SM Line on page 400
S0 Line on page 408

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Line

ACO Line
This type of line is only supported in IP500 V2 systems configured for operation as an Avaya
Cloud Office™ ATA gateway. Refer to the Deploying an IP Office as an Avaya Cloud Office ATA
Gateway manual.
Related links
Line on page 290
ACO Line | ACO on page 292
ACO Line | VoIP on page 293
ACO Line | T38 Fax on page 296

ACO Line | ACO

Navigation: Line > ACO Line > ACO
Configuration Settings
These settings are mergeable with the exception of the Line Number setting. Changing the Line
Number setting requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249
Enter the line number that you wish. Note that this must be unique. This value is used as
the incoming group ID for the line needed for incoming call routing.
ACO Domain Name Default = Blank.
ACO Proxy These two values should be set to match the values provided for the customer by Avaya
Address Cloud Office™, omitting the port number shown in that information.
Outgoing Group ID Default = 96666
Fixed value. This value is used is used as the Line Group ID on short codes used to
route calls to Avaya Cloud Office™.
URI Type Default = SIP URI.
Set the format the IP Office uses for SIP URI entries in headers.
• SIP URI - Use SIP URI format. For example, display <sip:content@hostname>
• Tel - Use Tel URI format. For example, +1-425-555-4567. This affects the From field
of outgoing calls. The To field for outgoing calls uses the format specified by the short
codes used for outgoing call routing.
• SIPS - Use SIPS format for all URIs. SIPS can be used only when Layer 4 Protocol is
set to TLS.
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 ACO Line

Field Description
Location Default = Cloud.
You can set Location values for the IP Office system and for individual extensions and
lines. Associating a line with a location:
• Applies the location's call admission control (CAC) settings to the line. See Configuring
Call Admission Control on page 709.
• For SIP lines that support RFC4119/RFC5139, emergency calls using the line can
include the location's address information.
• For more information, see Using Locations on page 617.

Network Configuration
Field Description
Layer 4 Protocol Default = TLS
Fixed value. Not changeable.
Use Network Default = None.
Topology Info
• LAN1 - Associate the line with the Network Topology and DiffServ Settings settings
of IP Office LAN1.
- If no STUN server address is set for the LAN interface, then the Binding Refresh
Time is ignored when calculating the timing for periodic OPTIONS messages unless
the Firewall/NAT Type is set to Open Internet.
• LAN2 - As above but using the settings of IP Office LAN2.
• None - If selected, the IP Office does not apply STUN lookup. The IP Office system IP
routing tables determine routing for the line.
Send Port Default = 5096
Fixed value. Not changeable.
Listen Port Default = 5061
Fixed value. Not changeable.

Related links
ACO Line on page 292

ACO Line | VoIP

Navigation: Line > ACO Line > VoIP
This form is used to configure the VoIP settings applied to calls on the ACO line.
You can merge these settings without needing to reboot the IP Office.

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Line

Configuration Settings
Field Description
Re-Invite Default = Off.
Supported
When enabled, the IP Office can use Re-Invite during a call to change the
characteristics of the call. For example, when the target of an incoming call or a transfer
does not support the codec originally negotiated on the trunk.
• Requires the ITSP to also support Re-Invite.
• This setting must be enabled for video support.
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the order
of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
Fax Transport Default = None.
Support
This option is available only if Re-Invite Supported is selected.
• IP500 V2 systems can terminate T38 fax calls.
• Linux-based IP Office systems can route the calls between trunks/terminals with
compatible fax types.
• Set the method the IP Office uses to handle fax calls.
The supported options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does not
support T38, the IP Office will send a re-invite to change the transport method to
G.711.
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 ACO Line

Field Description
Call Initiation Default = 4 seconds. Range = 1 to 99 seconds.
Timeout (s)
Sets how long the IP Office system should wait for a response to an attempt to initiate a
call before following the alternate routes set in an ARS form.
DTMF Support Default = RFC2833 (IP500 V2), RFC2833/RFC4733 (Linux-Based Server)
Selects the method the IP Office uses to signal DTMF key press digits to the remote end.
The options are:
• In Band - Send digits as part of the audio path.
• RFC2833 or RFC2833/RF4733 - Send digits using a separate audio stream from the
voice path. If not supported by the far end, the line reverts to using In Band signaling.
• Info - Send the digits in SIP INFO packets.
Media Security Default = Enforced.
These setting control whether SRTP is used for this line and the settings used for the
SRTP. The options are:
• Same as System: Matches the system setting at System | System | VoIP | VoIP
Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data) is
enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on System
| System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be protected
using encryption. The default is to encrypt just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be protected
using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.

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Line

Related links
ACO Line on page 292

ACO Line | T38 Fax

Navigation: Line > ACO Line > T38 FAX
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which they
both support. The options are: 0, 1, 2, 3.
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For UDPTL,
redundancy error correction is supported. Forward Error Correction (FEC) is not
supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed V.21
T.30 fax transmissions.
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27 and
V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax equipment or is
found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Default = 150.
Timeout (secs)
Scan Line Fix-up Default = On.
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 Analog Line

Field Description
TFOP Default = On.
Enhancement
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression
NSF Override Default = Off.
If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be
overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
ACO Line on page 292

Analog Line
Analog trunks can be provided within the systems in the following ways. In all cases the physical
ports are labeled as Analog. For full details of installation refer to the IP Office Installation manual.
Using ICLID: The system can route incoming calls using the ICLID received with the call.
However ICLID is not sent instantaneously. On analog trunks set to Loop Start ICLID, there will be
a short delay while the system waits for any ICLID digits before it can determine where to present
the call.
Line Status: Analog line do not indicate call status other than whether the line is free or in use.
Some system features, for example retrieving unanswered forwards and making twinned calls
make use of the call status indicated by digital lines. This is not possible with analog lines. Once
an analog line has been seized, the system has to assume that the call is connected and treats it
as having been answered.
Dialing Complete: The majority of North-American telephony services use en-bloc dialing.
Therefore the use of a ; is recommended at the end of all dialing short codes that use an N.
This is also recommended for all dialing where secondary dial tone short codes are being used.
Ground Start: This type of analog trunk is only supported through the Analog Trunk external
expansion module.
Related links
Line on page 290
Line Settings on page 298
Analog Options on page 299

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Line

Line Settings
Navigation: Line | Analog Line | Line Settings
Configuration Settings
These settings are mergeable with the exception of the Network Type setting. Changes to this
setting will require a reboot of the system.
Field Description
Line Number This parameter is not configurable, it is allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line. For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks on
the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
Incoming Group ID Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs is used to
match it to incoming call routes in the system configuration. The matching incoming call
route is then used to route incoming calls. The same ID can be used for multiple lines.
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 Analog Line

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Outgoing Channels Default = 1 (not changeable)
Voice Channels Default = 1 (not changeable)
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if all users
must dial a prefix to access an outside line. The prefix is automatically placed in front of
all incoming numbers so that users can dial the number back.
For outgoing calls: The system does not strip the prefix, therefore any prefixes not
suitable for external line presentation should be stripped using short codes.
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits. Allows a number to be assigned to the
ID line to identify it. On phones that support call appearance buttons, a Line Appearance
button with the same number will show the status of the line and can be used to answer
calls on the line. The line appearance ID must be unique and not match any extension
number.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.

Related links
Analog Line on page 297

Analog Options
Navigation: Line | Analog Line | Analog Options

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Line

Covers analog line specific settings. The system wide setting System | Telephony | Tones &
Music | CLI Type is used for to set the incoming CLI detection method for all analogue trunks.
The Allow Analog Trunk to Trunk Connect setting is mergeable. The remaining settings are not
mergeable. Changes to these settings will require a reboot of the system.
Field Description
Channel Set by the system. Shown for information only.
Trunk Type Default = Loop Start
Sets the analog line type. The options are:
• Ground Start: Ground Start is only supported on trunks provided by the Analog
Trunk 16 expansion module. It requires that the module and the control unit are
grounded. Refer to the IP Office installation manual.
• Loop Start
• Loop Start ICLID: As the system can use ICLID to route incoming calls, on
analog Loop Start ICLID trunks there is a few seconds delay while ICLID is
received before the call routing can be determined.
Signaling Type Default = DTMF Dialing
Sets the signaling method used on the line. The options are: DTMF Dialing, Pulse
Dialing.
Direction Default = Both Directions
Sets the allowed direction of operation of the line. The options are: Incoming,
Outgoing, Both Directions.
Flash Pulse Width Default = 0. Range = 0 to 2550ms.
Set the time interval for the flash pulse width.
Await Dial Tone Default = 0. Range = 0 to 25500ms.
Sets how long the system should wait before dialing out.
Echo Cancellation Default = 16ms.
The echo cancellation should only be adjusted as high as required to remove echo
problems. Setting it to a higher value than necessary can cause other distortions.
Not used with external expansion module trunks. The options are (milliseconds):
Off, 8, 16, 32, 64, 128.
Echo Reduction Default = On. (ATM4Uv2 card only)
Used when impedance matching is not required but echo reduction is.
Mains Hum Filter Default = Off.
If mains hum interference on the lines is detected or suspected, this settings can be
used to attempt to remove that interference. Useable with ATM16 trunks and IP500
ATM4U trunks. The options are: Off, 50Hz, 60Hz.
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 Analog Line

Field Description
Impedance Set the impedance used for the line. This field is only available for system locales
where the default value can be changed.
The value used for Default is set by the setting System | System | Locale. For
information, see Avaya IP Office Locale Settings.
The following values are used for Automatic Impedance Matching: 600+2150nF,
600, 900+2150nF, 900, 220+820||115nF, 370+620||310nF, 270+750||150nF,
320+1050||230nF, 350+1000||210nF, 800+100||210nF.
Quiet Line This field is only available for certain system locales (see above). The setting may
be required to compensate for signal loss on long lines.
Digits to break dial tone Default = 2. Range = Up to 3 digits.
During automatic impedance testing (see below), once the system has seized a
line, it dials this digit or digits to the line. In some cases it may be necessary to use
a different digit or digits. For example, if analog trunk go via another PBX system
or Centrex, it will be necessary to use the external trunk dialing prefix of the remote
system plus another digit, for example 92.
Automatic Default = Yes. (ATM4Uv2 card only)
When set to Yes, the Default value is used. The value used for Default is set by the
system Locale.
When set to No, the Impedance value can be manually selected from the list of
possible values:
• 600
• 900 270+(750R || 150nF) and 275R + (780R || 150nF)
• 220+(820R || 120nF) and 220R+ (82R || 115nF)
• 370+(620R || 310nF)
• 320+(1050R || 230nF)
• 370+(820R || 110nF)
• 275+(780R || 115nF)
• 120+(820R || 110nF)
• 350+(1000R || 210nF)
• 200+(680R || 100nF)
• 600+2.16μF
• 900+1μF
• 900+2.16μF
• 600+1μF Global Impedance
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Line

Field Description
Automatic Balance These controls can be used to test the impedance of a line and to then display
Impedance Match the best match resulting from the test. Testing should be performed with the line
connected but the system otherwise idle. To start testing click Start. The system
will then send a series of signals to the line and monitor the response, repeating
this at each possible impedance setting. Testing can be stopped at any time by
clicking Stop. When testing is complete, Manager will display the best match and
ask whether that match should be used for the line. If Yes is selected, Manager will
also ask whether the match should be applied to all other analog lines provided by
the same analog trunk card or module.
Note that on the Analog Trunk Module (ATM16), there are four control devices, each
supporting four channels. The impedance is set by the control device for all four
channels under its control. Consequently, the impedance match tool only functions
on lines 1, 5, 9, and 13.
Before testing, ensure that the following system settings are correctly set:
• System | System | Locale
• System | Telephony | Telephony | Companding Law
If either needs to be changed, make the required change and save the setting to the
system before proceeding with impedance matching.
Due to hardware differences, the impedance matching result will vary slightly
depending on which type of trunk card or expansion module is being used.
Automatic Balance Impedance Matching, Quiet Line and Digits to break dial
tone are available for the Bahrain, Egypt, French Canadian, India, Kuwait, Morocco,
Oman, Pakistan, Qatar, Saudi Arabia, South Africa, Turkey, United Arab Emirates,
United States and Customize locales.
Allow Analog Trunk to Default = Not selected (Off). When not enabled, users cannot transfer or forward
Trunk Connect external calls back off-switch using an analog trunk if the call was originally made
or received on another analog trunk. This prevents transfers to trunks that do not
support disconnect clear.
If the setting System | Telephony | Telephony | Unsupervised Analog Trunk
Disconnect Handling is enabled, this setting is greyed out and trunk to trunk
connections to any analog trunks are not allowed.
BCC Default = Not selected [Brazil locale only]
A collect call is a call at the receiver's expense and by his permission. If supported
by the line provider, BCC (Block Collect Call) can be used to bar collect calls.
Long CLI Line Default = Off
The CLI signal on some analog lines can become degraded and is not then
correctly detected. If you are sure that CLI is being provided but not detected,
selecting this option may resolve the problem.
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 Analog Line

Field Description
Modem Enabled Default = Off
The first analog trunk in a control unit can be set to modem operation (V32 with
V42 error correction). This allows the trunk to answer incoming modem calls and
be used for system maintenance. When on, the trunk can only be used for analog
modem calls. The default system short code *9000* can be used to toggle this
setting.
For the IP500 ATM4U-V2 Trunk Card Modem, it is not required to switch the card's
modem port on/off. The trunk card's V32 modem function can be accessed simply
by routing a modem call to the RAS service's extension number. The modem call
does not have to use the first analog trunk, instead the port remains available for
voice calls.
MWI Standard Default = None.
This setting is only displayed for ATM4U-V2 cards. When System | Voicemail |
Voicemail Type is set to Analogue MWI, change this setting to Bellcore FSK.
BCC Flash Pulse Width Default = 100 (1000ms). Range = 0 to 255.
Brazil locale only. Sets the BCC (Block collect call) flash pulse width.

Pulse Dialing
These settings are used for pulse dialing.
Field Description
Mark Default = 40ms. Range = 0 to 255.
Interval when DTMF signal is kept active during transmission of DTMF signals.
Space Default = 60ms. Range = 0 to 255.
Interval of silence between DTMF signal transmissions.
Inter-Digit Pause Default = 500ms. Range = 0 to 2550ms.
Sets the pause between digits transmitted to the line.

Ring Detection
These settings are used for ring detection.
Field Description
Ring Persistency Default = Set according to system locale. Range = 0 to 2550ms.
The minimum duration of signal required to be recognized.
Ring Off Maximum Default = Set according to system locale. Range = 0 to 25500ms.
The time required before signaling is regarded as ended.

Disconnect Clear
Disconnect clear (also known as 'Line Break' or 'Reliable Disconnect') is a method used to signal
from the line provider that the call has cleared. The system also uses 'Tone Disconnect', which

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Line

clears an analog call after 6 seconds of continuous tone, configured through the Busy Tone
Detection (System | Telephony | Tones & Music) settings.
Field Description
Disconnect Clear Default = On
Enables the use of disconnect clear.
If the setting System | Telephony | Telephony | Unsupervised Analog Trunk
Disconnect Handling is enabled, this setting is greyed out and disconnect clear
disabled.
Units Default = 500ms. Range = 0 to 2550ms.
This time must be less than the actual disconnect time period used by the line
provider by at least 150ms.

Secondary Dial Tone

Configures the use of secondary dial tone on analog lines. This is a different mechanism from
secondary dial tone using short codes. This method is used mainly within the Russian locale.
When selected, the options are:
Field Description
Secondary Dial Tone Default = Off
Await time: Default = 3000ms. Range = 0 to 25500ms.
Used when secondary dial tone (above) is selected. Sets the delay.
After n Digits Default = 1. Range = 0 to 10.
Sets where in the dialing string, the delay for secondary dial tone, should occur.
Matching Digit Default =8. Range = 0 to 9.
The digit which, when first matched in the dialing string, will cause secondary dial
tone delay.

DTMF
These settings are used for DTMF dialing.
Field Description
On Default = 80ms. Range = 0 to 255ms.
The width of the on pulses generated during DTMF dialing.
Off Default = 80ms. Range = 0 to 255ms.
The width of the off pulses generated during DTMF dialing.

Gains
These settings are used to adjust the perceived volume on all calls.

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 BRI Line

Field Description
A|D Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps.
Sets the analog to digital gain applied to the signal received from the trunk by
the system. To conform with the Receive Objective Loudness Rating at distances
greater than 2.7km from the central office, on analog trunks a receive gain of 1.5dB
must be set.
D|A Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps.
Sets the digital to analog gain applied to the signal from the system to the trunk.
Voice Recording Default = Low
Used to adjust the volume level of calls recorded by voicemail. The options are
Low, Medium or High.

Related links
Analog Line on page 297

BRI Line
BRI trunks are provided by the installation of a BRI trunk card into the control unit. The cards are
available in different variants with either 2 or 4 physical ports. Each port supports 2 B-channels for
calls. For full details of installation refer to the IP Office Installation manual.
Point-to-Point or Multipoint

BRI lines can be used in either Point-to-Point or Point-to-Multipoint mode. Point-to-Point lines are
used when only one device terminates a line in a customer's office. Point-to-Multipoint lines are
used when more than one device may be used on the line at the customer's premises. There are
major benefits in using Point-to-Point lines:-
• The exchange knows when the line/terminal equipment is down/dead, thus it will not offer
calls down that line. If the lines are Point-to-Multipoint, calls are always offered down the
line and fail if there is no response from the terminal equipment. So if you have two Point-to-
Multipoint lines and one is faulty 50% of incoming calls fail.
• You get a green LED on the Control Unit when the line is connected. With Point-to-Multipoint
lines some exchanges will drop layer 1/2 signals when the line is idle for a period.
• The timing clock is locked to the exchange. If layer 1/2 signals disappear on a line then the
Control Unit will switch to another line, however this may result in some audible click when
the switchover occurs.
The system's default Terminal Equipment Identifier (TEI) will normally allow it to work on Point-to-
Point or Point-to-Multipoint lines. However if you intend to connect multiple devices simultaneously
to an BRI line, then the TEI should be set to 127. With a TEI of 127, the control unit will ask the
exchange to allocate a TEI for operation.

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Line

Note:
When connected to some manufactures equipment, which provides an S0 interface (BRI), a
defaulted Control Unit will not bring up the ISDN line. Configuring the Control Unit to a TEI of
127 for that line will usually resolve this.
Related links
Line on page 290
BRI Line on page 306
Channels on page 310

BRI Line
Navigation: Line | BRI Line
The following settings are not mergeable. Changes to these settings will require a reboot of the
system.
• Line Sub Type, Network Type, TEI, Add 'Not-end-to-end ISDN' Information Element,
Progress Replacement, Clock Quality, Force Number Plan to ISDN, Number of
Channels.
Decreasing the Number of Channels setting requires a “merge with service disruption”. When the
configuration file is sent to the system, active calls on the deleted channels are cleared.
The remaining settings are mergeable.
Field Description
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Line Number This parameter is not configurable; it is allocated by the system.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Line Sub Type Default = NTT for Japan/ETSI for other locales.
Select to match the particular line type provided by the line provider. IP500 BRI daughter
cards can be configured for S-Bus (So) operation for connection to ISDN terminal
devices. Note that this requires the addition of terminating resistors at both the system
and remote ends, and the use of a suitable cross-over cable. For full details refer to the
Deploying Avaya IP Office Platform IP500 V2 manual.
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 BRI Line

Field Description
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Prefix Default = Blank. The prefix is used in the following ways:
• For incoming calls: The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls: The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
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Line

Field Description
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a number
is presented from ISDN as an "international number" this prefix is added. For example
441923000000 is converted to 00441923000000.
TEI Default = 0 The Terminal Equipment Identifier. Used to identify each device connected
to a particular ISDN line. For Point-to-Point lines this is 0. It can also be 0 on a Point to
Multipoint line, however if multiple devices are sharing a Point-to-Multipoint line it should
be set to 127 which results in the exchange allocating the TEI's to be used.
Number of Default = 2. Range = 0 to 2.
Channels
Defines the number of operational channels that are available on this line.
Outgoing Channels Default = 2. Range = 0 to 2.
This defines the number of channels available, on this line, for outgoing calls. This should
normally be the same as Number of Channels field, but can be reduced to ensure
incoming calls cannot be blocked by outgoing calls.
Voice Channels Default = 2. Range = 0 to 2.
The number of channels available for voice use.
Data Channels Default = 2. Range = 0 to 2.
The number of channels available for data use. If left blank, the value is 0.
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from this
line. Preference should always be given to using the clock source from a central office
exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available, Fallback
can be used to specify a clock source to use should the Network source not be
available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
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 BRI Line

Field Description
Add 'Not-end-to- Default = Never*. Sets whether the optional 'Not end-to-end ISDN' information element
end ISDN' should be added to outgoing calls on the line. The options are Never, Always or POTS
Information (only if the call was originated by an analog extension). *The default is Never except
Element for the following locales; for Italy the default is POTS, for New Zealand the default is
Always.
Progress Default = None.
Replacement
Progress messages are defined in the Q.931 ISDN connection control signaling protocol.
Generally, if a progress message is sent, the caller does not get connected and so
typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages. The
options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not hear
the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
Supports Partial Default = Off.
Rerouting
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network and
QSIG) ISDN exchange calls. When an external call is transferred to another external
number, the transfer is performed by the ISDN exchange and the channels to the system
are freed. Use of this service may need to be requested from the line provider and may
incur a charge.
Force Number Plan Default = Off.
to ISDN
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from Unknown/
Unknown to ISDN/Unknown.
Send Redirecting Default = Off.
Number
This option can be used on ISDN trunks where the redirecting service is supported by
the trunk provider. Where supported, on twinned calls the caller ID of the original call is
passed through to the twinning destination. This option is only used for twinned calls.
Support Call Default = Off. The system supports the triggering of malicious caller ID (MCID) tracing at
Tracing the ISDN exchange. Use of this feature requires liaison with the ISDN service provider
and the appropriate legal authorities to whom the call trace will be passed. The user
will also need to be enabled for call tracing and be provider with either a short code or
programmable button to activate MCID call trace. Refer to Malicious Call Tracing in the
Telephone Features section for full details.
Active CCBS Default = Off.
Support
Call completion to a busy subscriber (CCBS). It allows automatic callback to be used
on outgoing ISDN calls when the destination is busy. This feature can only be used on
point-to-point trunks. Use of this service may need to be requested from the line provider
and may incur a charge.
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Line

Field Description
Passive CCBS Default = Off.
Cost Per Charging The information is provided in the form of charge units. This setting is used to enter
Unit the call cost per charging unit set by the line provider. The values are 1/10,000th of a
currency unit. For example if the call cost per unit is £1.07, a value of 10700 should be
set on the line. Refer to Advice of Charge.
Send original Default = Off.
calling party for
Use the original calling party ID when forwarding calls or routing twinned calls.
forwarded and
twinning calls This setting applies to BRI lines with subtype ETSI.
Originator number Default = blank.
for forwarded and
The number used as the calling party ID when forwarding calls or routing twinned calls.
twinning calls
This field is grayed out when the Send original calling party for forwarded and
twinning calls setting is enabled.
This setting applies to BRI lines with subtype ETSI.

Related links
BRI Line on page 305

Channels
Navigation: Line | BRI Line | Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift
and then click Edit. When editing multiple channels, fields that must be unique such as Line
Appearance ID are not shown.
These settings are mergeable. Changes to these settings do not require a system reboot.
Field Description
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance ID
must be unique and not match any extension number. Line appearance is not supported
for trunks set to QSIG operation and is not recommended for trunks be used for DID.

Related links
BRI Line on page 305

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 H.323 Line

H.323 Line
These lines are added manually. They allow voice calls to be routed over data links within
the system. They are therefore dependent on the IP data routing between the system and the
destination having being configured and tested.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
Network Assessments
Not all data connections are suitable for voice traffic. A network assessment is required for internal
network connections. For external network connections a service level agreement is required from
the service provider. Avaya cannot control or be held accountable for the suitability of a data
connection for carrying voice traffic.
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice Networking
licenses.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Related links
Line on page 290
VoIP Line on page 311
Short Codes on page 313
VoIP Settings on page 314

VoIP Line
Navigation: Line | H.323 Line | VoIP Line
Configuration Settings
These settings are mergeable. Changes to these settings does not require a reboot of the system.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
Telephone Number Used to remember the telephone number of this line. For information only.
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Field Description
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or
Private.
• The IP Office will return number busy indication to any attempt to connect a call
on a Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using
any of the following features: multi-site networks, VPNremote, application
telecommuter mode.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added.
For example 441923000000 is converted to 00441923000000.
Location Default = Cloud.
You can set Location values for the IP Office system and for individual extensions
and lines. Associating a line with a location:
• Applies the location's call admission control (CAC) settings to the line. See
Configuring Call Admission Control on page 709.
• For SIP lines that support RFC4119/RFC5139, emergency calls using the line can
include the location's address information.
• For more information, see Using Locations on page 617.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The
description is not used elsewhere.
Send original calling Default = Off.
party for forwarded and
Use the original calling party ID when forwarding calls or routing twinned calls.
twinning calls
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 H.323 Line

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available
line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved
for the SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in the
network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Number of Channels Default = 20, Range 1 to 250.
Defines the number of operational channels that are available on this line.
Outgoing Channels Default = 20, Range 0 to 250.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
TEI Default = 0. Range = 0 to 127.
The Terminal Equipment Identifier. Used to identify each Control Unit connected to
a particular ISDN line. For Point to Point lines this is typically (always) 0. It can also
be 0 on a Point to Multi-Point line, however if multiple devices are actually sharing
a Point to Multi-Point line it should be set to 127 which will result in the exchange
deciding on the TEI's to be used by this Control Unit.

Related links
H.323 Line on page 311

Short Codes
Navigation: Line | H.323 Line | Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those

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Line

types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Changes to these settings do not require a reboot of the system.
Related links
H.323 Line on page 311

VoIP Settings
Navigation: Line | H.323 Line | VoIP Settings
This form is used to configure the VoIP setting applied to calls on the H.323 line.
Configuration Settings
These settings are mergeable. Changes to these settings does not require a reboot of the system.
Field Description
Gateway IP Address Default = Blank
Enter the IP address of the gateway device at the remote end.
Port Default = 1720
The H.323 line is identified by the IP Address:Port value. Specifying a unique port
value for this IP address allows multiple lines to use the same IP address.
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 H.323 Line

Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
Supplementary Default = H450.
Services
Selects the supplementary service signaling method for use across the H.323 trunk.
The remote end of the trunk must support the same option. The options are:
• None: No supplementary services are supported.
• H450: Use for H.323 lines connected to another PBX or device that uses H450.
• QSIG: Use for H.323 lines connected to another PBX or device that uses QSIG.
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
VoIP Silence Default = Off.
Suppression
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems, the
same setting should be set at both ends.
Enable FastStart for Default = Off
non-Avaya IP Phones
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
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Line

Field Description
Fax Transport Support Default = Off
This option is only supported on trunks with their Supplementary Services set
to IP Office SCN or IP Office Small Community Network - Fallback. Fax relay
is supported across H.323 multi-site network lines with Fax Transport Support
selected. This will use 2 VCM channels in each of the systems. Fax relay is only
supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition Linux servers.
Local Tones Default = Off
When selected, the tones are generated by the local system to which the phone
is registered. This option should not be used with lines being used for a multi-site
network.
DTMF Support Default = Out of Band
DTMF tones can be sent to the remote end either as DTMF tones within the
calls audio path (In Band) or a separate signals (Out of Band). Out of Band
is recommended for compression modes such as G.729 and G.723 compression
modes where DTMF in the voice stream could become distorted.
Allow Direct Media Path Default = On
This settings controls whether calls between IP endpoints and/or lines must go
through the IP Office or can try to route directly if possible within the customer
network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a
voice compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both
ends must support direct media and have matching VoIP settings, for example
using the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so
on. Otherwise, the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.
Progress Ends Overlap Default = Off.
Send
Some telephony equipment, primarily AT&T switches, over IP trunks send a
H.323 Progress rather than H.323 Proceeding message to signal that they have
recognized the digits sent in overlap state. By default the system expects an H.323
Proceeding message. This option is not available by default. If required, the value
ProgressEndsOverlapSend must be entered into the Source Numbers tab of the
NoUser user.
Default Name From Default = Off.
Display IE
When set, the Display IE is used as the default source for the name.

Related links
H.323 Line on page 311

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 IP DECT Line

IP DECT Line
This type of line can be manually added. They are used to route voice calls over an IP data
connection to an Avaya IP DECT system. Only one IP DECT line can be added to a system. Refer
to the IP DECT R4 Installation manual for full details.
Currently, only one IP DECT line is supported on a system.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Related links
Line on page 290
Line | IP DECT Line on page 317
Gateway on page 317
VoIP on page 320

Line | IP DECT Line

Navigation: Line | IP DECT Line | Line
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Line Number This number is allocated by the system and is not adjustable.
Associated Extensions Lists all the DECT extensions associated with the IP DECT line by the extension's
DECT Line ID setting.
Call based Location If enabled, the DECT extension location can be overridden on a call-by-call basis
Information using the location specified in the base station configuration. Supported with R11.1
FP2 SP2 and higher. Requires each base station to be configured with a location ID
that matches a location in the IP Office configuration. Refer to the IP Office DECT
R4 Installation manual.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The
description is not used elsewhere.

Related links
IP DECT Line on page 317

Gateway
Navigation: Line | IP DECT Line | Gateway
This form is used to configure aspects of information exchange between the IP Office and IP
DECT systems.

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Line

When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Auto-Create Default = Off.
Extension
If enabled, subscription of a handset with the DECT system causes the auto-creation of
a matching numbered extension within the system configuration if one does not already
exist. This setting is not supported on systems configured to use WebLM server licensing.
For security, auto-create is automatically disabled after 24 hours.
Auto-Create User Default = Off.
This option is only usable if Auto-Create Extension is also enabled. If enabled,
subscription of a handset with the DECT system causes the auto-creation of a matching
user within the system configuration if one does not already exist.
For security, any auto-create settings set to On are automatically set to Off after 24 hours.
Enable DHCP Default = Off
Support
This option is not supported for use with Avaya IP DECT R4. The IP DECT base stations
require DHCP and TFTP support. Enable this option if the system is being used to
provide that support, using IP addresses from its DHCP range (LAN1 or LAN2) and its
TFTP server setting. If not enabled, alternate DHCP and TFTP options must be provided
during the IP DECT installation.
• If it is desired to use the system for DHCP support of the ADMM and IP DECT
base stations only, the system address range should be set to match that number of
addresses. Those addresses are then taken during the system restart and will not be
available for other DHCP responses following the restart.
• For larger IP DECT installations, the use of a non-embedded TFTP software option
other than Manager is recommended.
Boot File Default = ADMM_RFP_1_0_0.tftp. Range = Up to 31 characters.
The name and path of the ADMM software file. The path is relative to the TFTP server
root directory.
ADMM MAC Default = 00:00:00:00:00:00
Address
This field must be used to indicate the MAC address of the IP DECT base station that
should load the ADMM software file and then act as the IP DECT system's ADMM. The
address is entered in hexadecimal format using comma, dash, colon or period separators.
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 IP DECT Line

Field Description
VLAN ID Default = Blank. Range = 0 to 4095.
If VLAN is being used by the IP DECT network, this field sets the VLAN address assigned
to the base stations by the system if Enable DHCP Support is selected.
• The system itself does not apply or use VLAN marking. It is assumed that the addition
of VLAN marking and routing of VLAN traffic is performed by other switches within the
customer network.
• An ID of zero is not recommended for normal VLAN operation.
• When blank, no VLAN option is sent to the IP DECT base station.
Base Station Default = Empty
Address List
This box is used to list the MAC addresses of the IP DECT base stations, other than
the base station being used as the ADMM and entered in the ADMM MAC Address
field. Right-click on the list to select Add or Delete. or use the Insert and Delete keys.
The addresses are entered in hexadecimal format using comma, dash, colon or period
separators.
Enable Provisioning
This option can be used with DECT R4 systems. It allows the setting of several values in the system
configuration that previously needed to be set separately in the master base stations configuration. For full
details refer to the DECT R4 Installation manual. The use of provisioning requires the system security settings
to include an IPDECT Group.
SARI/PARK Default = 0
Enter the PARK (Portable Access Rights Key) license key of the DECT R4 system. DECT
handset users enter this key when subscribing to the DECT system.
Subscriptions Default = Disabled
Select the method of subscription supported for handsets subscribing to the DECT R4
system. The options are:
• Disabled:Disables subscription of handsets.
• Auto-Create: Allow anonymous subscription of handsets. Once subscribed, the
handset is assigned a temporary extension number. That extension number can
be confirmed by dialing *#. A new extension number can be specified by dialing
<Extension Number>*<Login Code>#. The Auto-Create Extension and Auto-Create
User settings above should also be enabled. While configured to this mode, Manager
will not allow the manual addition of new IP DECT extensions.
• Preconfigured: Allow subscription only against existing IP DECT extensions records in
the system configuration. The handset IPEI number is used to match the subscribing
handset to a system extension.
Authentication Default = Blank.
Code
Set an authentication code that DECT handset users should enter when subscribing to
the DECT system.
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Field Description
Enable Resiliency
Default = Off.
Enables resiliency on the IP DECT Line. To configure resiliency, you must also configure an IP Office Line with
Backs up my IP Dect Phones set to On.
Status Enquiry Default = 30 seconds.
Period
The period between successive verifications on the H.323 channel. The smaller the
interval, the faster the IP DECT system recognizes that IP Office is down.
Prioritize Primary Default = Off.
Only available when Enable Provisioning is set to On.
Set to On for automatic fail-over recovery. When on, the IP DECT system switches
automatically from the backup IP Office to the "primary" IP Office.
Note that the IP DECT system does not switch back automatically from the backup
IP Office to the primary. The IP DECT system must be manually switched using Web
Manager.
Supervision Default = 120 seconds.
Timeout
Only available when Enable Provisioning is set to On.
The period of time the IP DECT system will wait between attempts to switch from the
backup IP Office to its "primary" IP Office.

Related links
IP DECT Line on page 317

VoIP
Navigation: Line | IP DECT Line | VoIP
Used to configure the VoIP setting applied to calls on the IP DECT line.
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Gateway IP Address Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Standby IP Address Default = Blank.
IP Address of the Standby Master IP Base Station or the second Mirror Base
Station. When the primary Mirror Base Station or Master Base Station is offline the
second Mirror or the Standby Master will take over and the system will use this IP
address.
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 IP DECT Line

Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend
all systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
VoIP Silence Default = Off.
Suppression
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
Allow Direct Media Path Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of
the calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the
media path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may
still allow calls between devices using the same audio codec to not require a
voice compression channel.

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Line

Related links
IP DECT Line on page 317

IP Office Line
This line type is used to connect two IP Office systems.
In previous releases, connecting two IP Office systems was achieved using H.323 Lines
configured with Supplementary Services set to IP Office SCN. In the current release, the IP
Office line type is used to connect IP Office systems. Separating out the IP Office line type
from the H.323 line type allows for the logical grouping of features and functions available when
connecting two IP Office systems, including IP Office systems connected through the cloud.
Note:
Setting an IP Office line with Transport Type = Proprietary and Networking Level = SCN
will interwork with a previous release system configured with an H.323 SCN line.
Related links
Line on page 290
Line on page 322
Short Codes on page 327
VoIP Settings on page 327
T38 Fax on page 330

Line
Navigation: Line | IP Office Line | Line
Additional configuration information
For information on the SCN Resiliency Options, see Server Edition Resiliency on page 806.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The
description is not used elsewhere.
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 IP Office Line

Field Description
Transport Type Default = Proprietary.
The options are
• Proprietary: The default connection type when connecting two IP Office
systems.
• WebSocket Client / Websocket Server: A WebSocket connection is an HTTP /
HTTPS initiated TCP pipe through which Call signalling and Network Signaling
is tunneled. This transport type is used to connect IP Office systems through the
cloud.
Selecting one of the WebSocket options enables the Security field and the
Password fields.
Networking Level Default = SCN.
The options are
• None: No supplementary services are supported.
• SCN: This option is used to link IP Office system within a multi-site network. The
systems within a multi-site network automatically exchange information about
users and extensions, allowing remote users to be called without any additional
configuration on the local system.
Security Default = Unsecured.
The Security field is available when Transport Type is set to WebSocket Client or
WebSocket Server.
The options are
• Unsecured : The connection uses HTTP/TCP.
• Medium: The connection uses HTTPS/TLS.
• High: The connection uses HTTPS/TLS. The server certificate store must contain
the client identity certificate.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or
Private.
• The IP Office will return number busy indication to any attempt to connect a call
on a Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using
any of the following features: multi-site networks, VPNremote, application
telecommuter mode.
Include location specific Default = Off.
information
Enabled when Network Type is set to Private. Set to On if the PBX on the other
end of the trunk is toll compliant.
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Line

Field Description
Telephone Number Default = Blank.
Used to remember the telephone number of this line. For information only.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available
line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved
for the SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in the
network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to
the secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to
the primary server.
Number of Channels Default = 20. Range 1 to 250; 1 to 500 for Select systems.
Defines the number of operational channels that are available on this line.
Outgoing Channels Default = 20, Range 0 to 250; 0 to 500 for Select systems.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.

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 IP Office Line

Gateway
Field Description
Address Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Location Default = Cloud.
You can set Location values for the IP Office system and for individual extensions
and lines. Associating a line with a location:
• Applies the location's call admission control (CAC) settings to the line. See
Configuring Call Admission Control on page 709.
• For SIP lines that support RFC4119/RFC5139, emergency calls using the line
can include the location's address information.
• For more information, see Using Locations on page 617.
Password Default = Blank.
Confirm Password The Password field is enabled when Transport Type is set to WebSocket Server
or WebSocket Client.
WebSockets are bi-directional HTTP or HTTPS communication pipes initiated from
a client to a server. They permit clients behind local a firewall to traverse the
internet to a server by using well known ports and protocols. A matching password
must be set at each end of the line.
Port When Transport Type is set to Proprietary, the default port is 1720 and cannot be
changed.
When Transport Type is set to WebSocket Client, the default port is 80.
The Port field is not available when Transport Type is set to WebSocket Server.
The HTTP and HTTPS receive ports are defined at the system level in the security
settings System Details tab.

SCN Resiliency Options

These options are only available when the Networking Level option is set to SCN. The intention
of this feature is to attempt to maintain a minimal level of operation while problems with the local
system are resolved.
For information on the SCN Resiliency Options, refer to the IP Office Resilience Overview
manual.
Field Description
Supports Resiliency Default = Off.
These fields are available when Networking Level is set to SCN. When selected,
all the available options are defaulted to On.
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Line

Field Description
Backs up my IP Phones Default = Off.
When selected, the local system shares information about the registered phones
and users on those phones with the backup system. If the local system is no longer
visible to the phones, the phones will reregister with the backup system. When
phones have registered with the backup system, they show an R on their display.
Note that while IP Office line settings are mergeable, changed to this setting require
the IP phones to be restarted in order to become aware of the change in their
failover destination.
If the setting System | Telephony | Telephony | Phone Failback is set to
Automatic, and the phone’s primary server has been up for more than 10 minutes,
the backup system causes idle phones to perform a failback recovery to the original
system.
If using resilience backup to support Avaya IP phones, Auto-create Extn and
Auto-create User should not be left enabled after initial configuration or any
subsequent addition of new extensions and users. Leaving auto-create options
enabled on a system that is a failover target may cause duplicate extension/user
records on the multi-site network under multiple failure scenarios.
Backs up my Hunt Default = Off.
Groups
This option is available only on the IP Office Line connecting the Server Edition
Primary server to the Server Edition Secondary server.
When selected, any hunt groups the local system is advertising to the network are
advertised from the backup system when fallback is required. The trigger for this
occurring is phones registered with the local system registering with the backup
system, ie. Backs up my IP Phones above must also be enabled.
When used, the only hunt group members that will be available are as follows:
• If the group was a distributed hunt group, those members who were remote
members on other systems are still visible within the network.
• Any local members who have hot desked to another system still visible within the
network.
When the local system becomes visible to the backup system again, the groups will
return to be advertised from the local system.
Backs up my Voicemail Default = Off.
This option can be used if the local system is hosting the Voicemail Pro server
being used by the network. If selected, when the local system is no longer visible
to the voicemail server, the backup system acts as host for the voicemail server.
In a Server Edition network, this option is only available on the H.323 trunk
from the Primary Server to the Secondary Server. It is assumed to be on and is
automatically set by the Resilience Administration tool.
The option requires the backup system to have licenses for the Voicemail Pro
features that are required to operate during any fallback period.
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Field Description
Backs up my IP DECT Default = Off.
Phones
This option is used for Avaya IP DECT phones registered with the system. When
selected, it will share information about the registered phones and users on those
phones with the backup system.
If the local system is no longer visible to the phones, the phones will reregister
with the backup system. The users who were currently on those phones will appear
on the backup system as if they had hot desked. Note that when the local system
is restored to the network, the phones will not automatically re-register with it. A
phone reset via either a phone power cycle or using the System Status Application
is required. When phones have registered with the backup system, they will show
an R on their display.

Note:
Only one IP Office Line can have this configuration parameter set to On.
Backs up my one-X Default = Off.
Portal
This option is available on Server Edition Select deployments and only on the
IP Office Line connecting the Server Edition Primary server to the Server Edition
Secondary server.
When set to On, this setting enables one-X Portal resiliency and turns on the
backup one-X Portal on the Server Edition Secondary server.
Backs up my Default = Off
Conferences
This option is available on the line from the primary to secondary server in Linux-
based networks. If enabled, the secondary server will provide hosting for system
meet-me conferences if the primary is not available.

Related links
IP Office Line on page 322

Short Codes
Navigation: Line | IP Office Line | Short Codes
Incoming calls on IP Office Lines are not routed using Incoming Call Route settings.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Related links
IP Office Line on page 322

VoIP Settings
Navigation: Line | IP Office Line | VoIP Settings

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Line

Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
Fax Transport Default = None.
Support
This option is available only if Re-Invite Supported is selected.
• IP500 V2 systems can terminate T38 fax calls.
• Linux-based IP Office systems can route the calls between trunks/terminals with
compatible fax types.
• Set the method the IP Office uses to handle fax calls.
The supported options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does not
support T38, the IP Office will send a re-invite to change the transport method to
G.711.
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
(s)
Sets how long the IP Office system should wait for a response to an attempt to initiate
a call before following the alternate routes set in an ARS form.
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Field Description
Media Security Default = Same as System.
Secure RTP (SRTP) can be used between IP Offices to add additional security. These
settings control whether SRTP is used for this line and the settings used for the SRTP.
The options are:
• Same as System: Matches the system setting at System | System | VoIP | VoIP
Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) is enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on
System | System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be protected
using encryption. The default is to encrypt just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be protected
using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any
audio data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
Out Of Band DTMF Default = On.
Out of Band DTMF is set to on and cannot be changed.
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Field Description
Allow Direct Media Default = On
Path
This settings controls whether calls between IP endpoints and/or lines must go
through the IP Office or can try to route directly if possible within the customer
network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a voice
compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both ends
must support direct media and have matching VoIP settings, for example using
the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on.
Otherwise, the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.

Related links
IP Office Line on page 322

T38 Fax
Navigation: Line | IP Office Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which they
both support. The options are: 0, 1, 2, 3.
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For UDPTL,
redundancy error correction is supported. Forward Error Correction (FEC) is not
supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed V.21
T.30 fax transmissions.
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Field Description
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27 and
V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax equipment or is
found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Default = 150.
Timeout (secs)
Scan Line Fix-up Default = On.
TFOP Default = On.
Enhancement
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression
NSF Override Default = Off.
If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be
overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
IP Office Line on page 322

Legacy SIP DECT Line

A Legacy SIP DECT Line can be added to connect to a D100 Base Station.
Related links
Line on page 290
SIP DECT Base on page 332

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VoIP on page 333

SIP DECT Base

Navigation: Line | Legacy SIP DECT Line | SIP DECT Base
The IP Office can support up to four D100 Base Stations. Each connects to the IP Office using a
Legacy SIP DECT Line.
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field Description
Line Number Default = Blank. A unique line number associated with the SIP DECT Base Station.
Associated Lists the SIP DECT extensions associated with the line through the extension's SIP DECT
Extensions Line setting.
Base Name Default = Blank. Maximum 16 characters.
A name assigned to the base station. Each base station provisioned on the IP Office must
have a unique name. The field cannot be blank. The format is an alphanumeric string with
no special characters.
Base MAC Default = Blank.
Address
The MAC Address of the base station. If only one base station is provisioned, the field can
remain at the default value. If multiple base stations are provisioned, the MAC address for
each base station must be entered.
Configure Base IP
Configure Base IP Default = Off.
Set to On to configure IP address attributes for the base station. When enabled, the
Configure Base IP settings are displayed.
DHCP Client Default = On.
When enabled, specifies that the base station operates as a DHCP client. When enabled,
not other IP address attributes can be configured.
IP Address Default = Blank.
The IP address of the base station. The IP address must be on the same subnet as one of
the LAN interfaces.
IP Mask Default = Blank.
IP address mask.
IP Gateway Default = Blank.
The default gateway address
Provisioning Default = IP Office interface address.
Server
The server address from where the Base Station configuration files can be retrieved.

Description Default = Blank. Maximum 31 characters.

You can use this field to enter a description for the configuration entry. The description is
not used elsewhere.

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 Legacy SIP DECT Line

Related links
Legacy SIP DECT Line on page 331

VoIP
Navigation: Line | Legacy SIP DECT Line | VoIP
This form is used to configure the VoIP setting applied to calls on a Legacy SIP DECT Line
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field Description
IP Address Default = Blank.
The IP address of the SIP DECT extension.
Codec Selection Default = Custom
This field defines the codec or codecs offered during call setup. The codecs available
to be used are set through System | System | VoIP | VoIP.
The Codec Selection option allows specific configuration of the codec preferences to
be different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs. The D100 Base Station supports only
G711 codecs.
TDM > IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP > TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
DTMF Support Default =RFC2833
The D100 Base Station supports only RFC2833.
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
Local Hold Music Default = Off
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Field Description
Allow Direct Media Default = On
Path
This settings controls whether calls between IP endpoints and/or lines must go
through the IP Office or can try to route directly if possible within the customer
network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a voice
compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both ends
must support direct media and have matching VoIP settings, for example using
the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on.
Otherwise, the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.
Reinvite Supported Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does
not support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite.

Related links
Legacy SIP DECT Line on page 331

MS Teams Line
IP Office can be configured as the telephony service for calls made to and from Microsoft Teams.
The MS Teams Line settings uses a private SIP trunk connection with Session Border Controller
(SBC).
Only one MS Teams line is supported, including for networked IP Office systems. For IP Office
Server Edition and Select, the line should be configured on the primary server.
For details, see the Deploying MS Teams Direct Routing with IP Office manual.
Related links
Line on page 290
MS Teams on page 334
VoIP on page 337
Engineering on page 341
SIP Credentials on page 342

MS Teams
Navigation: Line | MS Teams Line | MS Teams

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 MS Teams Line

Additional configuration information

For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 621.
Configuration settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Changing the In Service setting to Disabled (out of service) requires a system reboot. However,
changing the In Service setting to Enabled is mergeable. Configuration changes made while the
line is out of service are also mergeable.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
The line number must be unique. On IP500 V2 systems, line numbers 1 to 16 are
reserved for internal hardware.
• Only one MS Teams line is supported, including for networked IP Office systems.
For IP Office Server Edition and Select, the line should be configured on the
primary server.
In Service Default = Enabled
This option can be used to administratively disable the MS Teams Line. It does not
reflect the dynamic state of the line.
Calling Number Default = Clear
Verification
These settings configure the SIP trunks use of STIR protocols for calling
number verification. For more details, see SIP Calling Number Verification (STIR/
SHAKEN) on page 866.
• Incoming Calls Handling: Default = System.
Sets the defaults for which calls are accepted by the system based on the
authentication level of the call. This default can be overridden in the individual
line configuration.
- Allow All - Allow all calls regardless of authentication level. Note this can
include calls with no authentication level.
- Allow Validated - Only accept calls which are fully or semi-authenticated.
- Allow Not Failed - Accept all calls expect those that specifically failed
authentication. Note this can include calls with no authentication level.
Domain Name Default = Blank.
An IP address or SIP domain name as required by the service provider.
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Field Description
Local Domain Name Default = Blank.
An IP address or SIP domain name as required by the service provider.
When configured, the Local Domain Name value is used in the following:
• From and Contact headers
• PAI header, when the setting Line | SIP Line | Advanced | Use Domain for PAI
is checked
• Diversion header
If both the ITSP Domain Name and the Local Domain Name are configured,
then Local Domain takes precedence.
Local Domain Name is not used in the Remote Party ID header.
Proxy Address Default- Blank
Enter the proxy address to send the packet.
Example: ms-teams.com
Outgoing Group ID Default = 97777
This value is not changeable. It can be used by short codes to route calls to the
line.
Prefix Default = Blank
This prefix is added to any source number received with incoming calls.
Max Calls Default = 10
Sets the number of simultaneous calls allowed using this line.
URI Type Default = SIP.
When SIP or SIP URI is selected, the SIP URI format is used (for example,
[email protected]). This affects the From field of outgoing calls. The To field
for outgoing calls always uses the format specified by the short codes used for
outgoing call routing.
Recommendation: When SIP Secured URI is required, the URI Type should be set
to SIP URI.
SIP URI can be used only when Layer 4 Protocol is set to TLS.
Media Connection Default = Enabled.
Preservation
When enabled, the system attempts to maintain established calls despite brief
network failures. Call handling features are not available when a call is in a
preserved state. When the Media Connection Preservation setting is enabled,
it applies to Avaya H.323 phones that support connection preservation.
Location
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 MS Teams Line

Field Description
Network Configuration
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Layer 4 Protocol Default = TCP.
Send Port When Layer 4 Protocol is set to TLS, the default setting is 5061. When Layer 4
Protocol is set to TCP, the default setting is 5060.
Listen Port When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.
Use Network Topology Default = None.
Info
This field associates the line with the LAN interface System | LAN | Network
Topology settings. It also applies the System | LAN | VoIP | DiffServ Settings to
the outgoing traffic on the line. If None is selected, STUN lookup is not applied and
routing is determined by the system's routing tables.
If no STUN server address is set for the interface, then the System | LAN |
Network Topology | Binding Refresh Time is ignored by MS Teams Lines when
calculating the periodic OPTIONS timing unless the Firewall/NAT Type is set to
Open Internet.
Session Time (seconds) Default = 1200. Range = 90 to 64800
This field specifies the session expiry time. At the halfway point of the expiry time,
a session refresh message is sent. Setting the Session Time (seconds) to On
Demand disables the session timer.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The
description is not used elsewhere.

Related links
MS Teams Line on page 334

VoIP
Navigation: Line | MS Teams Line | VoIP
These settings are mergeable. Changes to these settings do not require a reboot of the system.
These settings can be edited online. Changes to these settings do not require a reboot of the
system.

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Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup.
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The codecs available in this form are set through the codec list and the System
Default settings are on System | System | VoIP | VoIP.
Within a network of systems, it is strongly recommended that all the systems and
the lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used
to select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
Fax Transport Support Default = None.
This option is available only if Re-Invite Supported is selected.
• IP500 V2 systems can terminate T38 fax calls.
• Linux-based IP Office systems can route the calls between trunks/terminals with
compatible fax types.
• Set the method the IP Office uses to handle fax calls.
The supported options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does
not support T38, the IP Office will send a re-invite to change the transport
method to G.711.
Call Initiation Timeout (s) Default = 4 seconds. Range = 1 to 99 seconds.
Sets how long the IP Office system should wait for a response to an attempt to
initiate a call before following the alternate routes set in an ARS form.
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Field Description
DTMF Support Default = RFC2833 (IP500 V2), RFC2833/RFC4733 (Linux-Based Server)
Selects the method the IP Office uses to signal DTMF key press digits to the
remote end. The options are:
• In Band - Send digits as part of the audio path.
• RFC2833 or RFC2833/RF4733 - Send digits using a separate audio stream from
the voice path. If not supported by the far end, the line reverts to using In Band
signaling.
• Info - Send the digits in SIP INFO packets.
Media Security Default = Same as System.
These setting controls and settings of SRTP that is used for the selected line. The
options are:
• Same as System: Matches the system setting at System | System | VoIP |
VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) is enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and
data) is enforced to use SRTP only. Selecting Enforced on a line or extension
that does not support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on
System | System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be
protected using encryption. The default is to encrypt just the RTP stream (the
speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be
protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
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Line

Field Description
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any
audio data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
Re-Invite Supported Default = Off.
When enabled, the IP Office can use Re-Invite during a call to change the
characteristics of the call. For example, when the target of an incoming call or a
transfer does not support the codec originally negotiated on the trunk.
• Requires the ITSP to also support Re-Invite.
• This setting must be enabled for video support.
Codec Lockdown Default = Off.
In response to a SIP offer with a list of codecs, some SIP user agents send a SDP
answer that also lists multiple codecs. The user agent can then switch to any of
those codecs during the session without requiring further negotiation. However, IP
Office does not support this, so loss of speech path occurs if the current codec
changes without renegotiation.
• If enabled, when the IP Office receives an SDP answer with multiple codecs from
its list of offered codecs, the IP Office sends a re-INVITE using just a single
codec from the list, and an SIP offer with just the single chosen codec.
• This option requires Re-Invite Supported enabled.
Allow Direct Media Path Default = On
This settings controls whether calls between IP endpoints and/or lines must go
through the IP Office or can try to route directly if possible within the customer
network.
• If disabled, calls go through the IP Office and use its resources. RTP relay
support may allow calls between devices using the same audio codec to not
require a voice compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both
ends must support direct media and have matching VoIP settings, for example
using the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so
on. Otherwise, the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt
direct media even if the other end has differing DTMF settings.
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 MS Teams Line

Field Description
Force direct media with Default = On
phones
When enabled, if an Avaya IP phone dials digits during a direct media call, the
IP Office changes the call to indirect media and sends the digits as RFC2833.
15-seconds after the last digit, the IP Office changes the call back to direct media.
• This setting is requires the line to have Re-Invite Supported and Allow Direct
Media Path enabled, and DTMF Support set to RFC2833/RF4733.
G.711 Fax ECAN Default = Off
When enabled, if the IP Office detects a fax call, it switches to G.711 with echo
cancellation (ECAN) based on the 'G.711 Fax ECAN field, NLP disabled, a fixed
jitter buffer, and silence suppression is disabled. You can use this to avoid an
ECAN mismatch with the trunk provider.
• This setting is only available on IP500 V2 systems when Fax Transport Support
is set to G.711 or T38 Fallback.

Related links
MS Teams Line on page 334

Engineering
Navigation: Line | MS Teams Line | Engineering
You can use this tab to enter commands that apply special features to the SIP line. The
commands are called SIP Line Custom (SLIC) strings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
reINVITE Codec Renegotiation
For R11.0 and higher, the IP Office supports codec renegotiation when a reINVITE is received.
See Codec selection on page 857.
You can use the following command to retain the pre-R11.0 behavior of no renegotiation. Note: On
existing IP Office systems upgraded to R11.0 or higher, this command is automatically added to all
existing SIP lines.
• SLIC_PREFER_EXISTING_CODEC
Calling Number Validation
You can use the following commands to control calling number validation. See SIP Calling Number
Verification (STIR/SHAKEN) on page 866.
• SLIC_STIR_REJECT_CODE=<n> where <n> is the response code sent for calls rejected by
the IP Office.
• SLIC_STIR_REJECT_STRING=<y> where <y> is the response string sent for calls rejected
by the IP Office.
• SLIC_STIR_ATTEST="<w>" where <w> is the name of the header the IP Office checks for
a call's authorization level.
• SLIC_STIR_CUSTOM=<z> where <z> value enables or disables various call features.

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Line

Server Name Identification (SNI)

The following SLIC codes can be used for SIP trunks using TLS. When used:
• On outgoing connections, the IP Office adds Server Name Indication (SNI) information to the
SAN field it sends.
• If the IP Office system's Received certificate checks (Telephony endpoints) settings is set
to Medium + Remote Checks or High + Remote Checks, then the SLIC value is also used
to validate the received certificates SAN.
The SLIC codes are:
• SLI_ADD_SIP_SAN=<X>
Use a SNI set to sip:<SNI> where the <SNI> value is taken from the existing IP Office SIP
line configuration based on the following values of <X> as below:
- D = Use the value of the SIP line's ITSP Domain Name setting (Line > SIP Line). For
example, for a SIP line with the ITSP Domain Name set to ipo.example.com, adding
SLIC_ADD_SIP_SAN=D sets the SNI added to sip:ipo.example.com.
- P = Use the value of the SIP line's configured ITSP Proxy Address setting (Line >
Transport > ). This option is only supported for a ITSP Proxy Address set to a single
address. For example: SLI_ADD_SIP_SAN=P
Keepalives
Supported with IP Office R11.1.3.1 and higher.
You can add SLIC_HNT_EMPTY_PACKET to have the SIP line send RTP packets with payload 20
(unassigned payload) and no data as keepalives. This overrides the default of send STUN packets
for keepalives.
Related links
MS Teams Line on page 334

SIP Credentials
Navigation: Line | MS Teams Line | SIP Credentials
These settings in the SIP Credentials tab are used to enter the ITSP username and password
for the SIP account with the ITSP. If you have several SIP accounts going to the same ITSP IP
address or domain name, you can enter up to 30 sets of ITSP account names and passwords on
this tab.
Use the Add, Remove, and Edit buttons to manage the set of credentials for the SIP trunk
accounts.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 PRI Trunks

Field Descriptions
Index This number is assigned automatically and cannot be edited. If the From field on the
SIP URI being used for the call is set to Use Authentication Name , the registration
field of the SIP URI indicates the index number of the SIP credentials to use for calls
by that SIP URI.
User Name This name must be unique and is used to identify the trunk. The name can include the
domain if necessary.
Authentication Name Default = Blank.
This field can be blank but must be completed if a Password is also specified. This
value is provided by the SIP ITSP. Depending on the settings on the Local URI tab
associated with the SIP call, it may also be used as the user part of the SIP URI. The
name can include the domain if necessary.
Contact Default = Blank.
This field is used to enter a contact and can include the domain if necessary.
Password Default = Blank.
This value is provided by the SIP ITSP. If a password is specified, the matching
Authentication Name must also be set.
Expiration (mins) Default = 60 minutes.
This setting defines how often registration with the SIP ITSP is required following any
previous registration.
Registration Default = On.
Required
If selected, the fields above above are used for registration when making calls. If
exported or imported as part of a trunk template.

Related links
MS Teams Line on page 334

PRI Trunks
PRI trunks are provided by the installation of a PRI trunk card into the control unit. avThe IP500
PRI-U trunk card can be configured (see below) to one of those line types. The cards are also
available with either 1 or 2 physical ports. The number of B-channels supported by each physical
port depends on the line type of the card.
• E1: 30 B-channels and 1 D-channel per port.
• T1: 24 B-channels per port.
• US PRI: 23 B-channels and 1 D-channel per port.
• E1-R2: 30 B-channels and 1 D-channel per port.

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Line

IP500 PRI-U Trunk Card Line Type

The IP500 PRI-U card can be configured to support either E1, T1 or E1-R2 PRI line types. To
select the line type required, right-click on the line in the group or navigation pane and select
Change Universal PRI Card Line Type.
The control unit supports 8 B-channels on each IP500 PRI-U card fitted. Additional B-channels
up to the full capacity of IP500 PRI-U ports installed require licenses added to the configuration.
D-channels are not affected by licensing.
• For ETSI and QSIG trunks, license instances are consumed by the number of calls in
progress on B-channels.
• For T1, E1R2 and ETSI CHI trunks, licenses instances are consumed by the channels set as
in service.
Related links
Line on page 290
E1 Line on page 344
E1 R2 Line on page 352
T1 Line on page 358
T1 PRI Line on page 363

E1 Line
Related links
PRI Trunks on page 343
E1 PRI Line on page 344
E1 Short Codes on page 350
E1 PRI Channels on page 350

E1 PRI Line
Navigation: Line | E1 PRI Line
The following settings are not mergeable. Changes to these settings require a system reboot.
• Line Sub Type
• Network Type
• TEI
• Channel Allocation
• CRC Checking
• Clock Quality
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement
• Force Number Plan to ISDN
• Line Signalling

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 PRI Trunks

Decreasing the Number of Channels setting requires a “merge with service disruption”. When the
configuration file is sent to the system, active calls on the deleted channels are cleared.
The remaining settings are mergeable.
Field Description
Line Number This parameter is not configurable; it is allocated by the system.
Line Sub Type Select to match the particular line type provided by the line provider. The options are:
• ETSI
• ETSI CHI
• QSIG A
• QSIG B
ETSI CHI is used to send the channel allocation ID (CHI) in the call setup signaling. This
is a request to use a particular B-channel rather than use any B-channel allocated by the
central office exchange.
QSIG trunks are not supported on IP500 V2 systems without IP500 Voice Networking
licenses.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
Channel Allocation Default = 30|1.
For lines set to ETSI CHI, this option allows the system to select the default order in
which channels should be used for outgoing calls. Typically this is set as the opposite of
the default order in which the central office exchange uses channels for incoming calls.
For lines set to the Line Sub Type of ETSI CHI, the Incoming Group ID is set as part of
the individual channel settings.
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Field Description
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a number
is presented from ISDN as an "international number" this prefix is added. For example
441923000000 is converted to 00441923000000.
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 PRI Trunks

Field Description
TEI Default = 0 The
Terminal Equipment Identifier. Used to identify each Control Unit connected to a
particular ISDN line. For Point to Point lines this is typically (always) 0. It can also be 0 on
a Point to Multi-Point line, however if multiple devices are sharing a Point to Multi-Point
line it should be set to 127 which results in the exchange deciding on the TEI's to be
used.
Number of Defines the number of operational channels that are available on this line. Up to 30 for
Channels E1 PRI, 23 for T1 PRI.
Outgoing Channels This defines the number of channels available, on this line, for outgoing calls. This should
normally be the same as Number of Channels field, but can be reduced to ensure
incoming calls cannot be blocked by outgoing calls. Only available when the Line Sub
Type is set to ETSI.
Voice Channels The number of channels available for voice use. Only available when the Line Sub Type
is set to ETSI.
Data Channels The number of channels available for data use. Only available when the Line Sub Type
is set to ETSI.
CRC Checking Default = On
Switches CRC on or off.
Line Signalling Default = CPE This option is not used for lines where the Line SubType is set to QSIG.
Select either CPE (customer premises equipment) or CO (central office). The CO feature
is intended to be used primarily as a testing aid. It allows PRI lines to be tested in a
back-to-back configuration, using crossover cables.
The CO feature operates on this line type by modifying the way in which incoming calls
are disconnected for system configuration in Brazil and Argentina. In these locales, the
CO setting uses Forced-Release instead of Clear-Back to disconnect incoming calls. The
Brazilian Double-Seizure mechanism, used to police Collect calls, is also disabled in CO
mode.
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from this
line. Preference should always be given to using the clock source from a central office
exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available, Fallback
can be used to specify a clock source to use should the Network source not be
available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
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Line

Field Description
Add 'Not-end-to- Default = Never
end ISDN'
Sets whether the optional 'Not end-to-end ISDN' information element should be added to
Information
outgoing calls on the line. The options are:
Element
• Never
• Always
• POTS(only if the call was originated by an analog extension).
The default is Never except for the following locales:
• for Italy the default is POTS.
• for New Zealand the default is Always.
Progress Default = None.
Replacement
Progress messages are defined in the Q.931 ISDN connection control signaling protocol.
Generally, if a progress message is sent, the caller does not get connected and so
typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages. The
options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not hear
the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
Supports Partial Default = Off.
Rerouting
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network and
QSIG) ISDN exchange calls. When an external call is transferred to another external
number, the transfer is performed by the ISDN exchange and the channels to the system
are freed. Use of this service may need to be requested from the line provider and may
incur a charge.
Force Number Plan Default = Off.
to ISDN
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from Unknown/
Unknown to ISDN/Unknown.
Send Redirecting Default = Off.
Number
This option can be used on ISDN trunks where the redirecting service is supported by
the trunk provider. Where supported, on twinned calls the caller ID of the original call is
passed through to the twinning destination. This option is only used for twinned calls.
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 PRI Trunks

Field Description
Support Call Default = Off.
Tracing
The system supports the triggering of malicious caller ID (MCID) tracing at the ISDN
exchange. Use of this feature requires liaison with the ISDN service provider and
the appropriate legal authorities to whom the call trace will be passed. The user will
also need to be enabled for call tracing and be provider with either a short code or
programmable button to activate MCID call trace. Refer to Malicious Call Tracing in the
Telephone Features section for full details.
Active CCBS Default = Off.
Support
Call completion to a busy subscriber (CCBS). It allows automatic callback to be used
on outgoing ISDN calls when the destination is busy. This feature can only be used on
point-to-point trunks. Use of this service may need to be requested from the line provider
and may incur a charge.
Passive CCBS Default = Off.
Cost Per Charging Advice of charge (AOC) information can be output in SMDR. The information is provided
Unit in the form of charge units. This setting is used to enter the call cost per charging unit
set by the line provider. The values are 1/10,000th of a currency unit. For example if the
call cost per unit is £1.07, a value of 10700 should be set on the line. See Advice of
Charge on page 616.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Send original Default = Off.
calling party for
Use the original calling party ID when forwarding calls or routing twinned calls.
forwarded and
twinning calls This setting applies to the following ISDN lines:
• PRI24 with subtypes: PRI, QSIGA, QSIGB, ETSI, ETSI CHI.
• PRI30 with subtypes: QSIGA, QSIGB, ETSI, ETSI CHI.
Originator number Default = blank.
for forwarded and
The number used as the calling party ID when forwarding calls or routing twinned calls.
twinning calls
This field is grayed out when the Send original calling party for forwarded and
twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes: PRI, QSIGA, QSIGB, ETSI, ETSI CHI.
• PRI30 with subtypes: QSIGA, QSIGB, ETSI, ETSI CHI.

The following fields are shown for a US T1 trunk card set to ETSI or QSIG operation. These cards
have the same settings E1 PRI trunk cards set to ETSI or QSIG but only support 23 channels.
These settings are not mergeable. Changing these settings requires a system reboot.

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Line

Field Description
CSU Operation Check this field to enable the T1 line to respond to loop-back requests from the
line.
Haul Length Default = 0-115 feet
Sets the line length to a specific distance.
Channel Unit Default = Foreign Exchange This field should be set to match the channel signaling
equipment provided by the Central Office. The options are Foreign Exchange,
Special Access or Normal.

Related links
E1 Line on page 344

E1 Short Codes
Navigation: Line | E1 Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those
types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Changes to these settings do not require a reboot of the system.
Related links
E1 Line on page 344

E1 PRI Channels
Navigation: Line | E1 PRI Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift
and then click Edit. When editing multiple channels, fields that must be unique such as Line
Appearance ID are not shown.
The following settings are mergeable:
• Line Appearance ID (ETSI, ETSI CHI)

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 PRI Trunks

• Admin (ETSI CHI)

• Incoming Group ID (ETSI CHI)
• Outgoing Group ID (ETSI CHI)
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance
ID must be unique and not match any extension number. Line appearance is not
supported for trunks set to QSIG operation and is not recommended for trunks be used
for DID.
If the trunk Line Sub Type is set to ETSI CHI, outgoing line appearance calls must use
the corresponding channel.

The following additional fields are shown for lines where the Line Sub Type is set to ETSI CHI.
Field Description
Incoming Group ID Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs is used to
match it to incoming call routes in the system configuration. The matching incoming call
route is then used to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Direction Default = Bothways
The direction of calls on the channel. The options are:Incoming, Outgoing, Bothways.
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Line

Field Description
Bearer Default = Any
The type of traffic carried by the channel. The options are: Voice, Data, Any.
Admin Default = Out of Service.
This field can be used to indicate whether the channel is in use or not. On trunks where
only a limited number of channels have been requested from the trunk provider (known
as sub-equipped trunks), those channels not provided should be set as Out of Service.
For channels that are available but are temporarily not being used select Maintenance.
Tx Gain Default = 0dB. Range = -10dBb to +5dB.
The transmit gain in dB.
Rx Gain Default = 0dB. Range = -10dBb to +5dB.
The receive gain in dB.

Related links
E1 Line on page 344

E1 R2 Line
Navigation: Line | E1–R2 Line
Related links
PRI Trunks on page 343
E1-R2 Options on page 352
E1-R2 Channels on page 354
E1 R2 MFC Group on page 356
E1-R2 Advanced on page 356

E1-R2 Options
Navigation: Line | E1–R2 Options
Changing the Admin setting is mergeable. The remaining settings are not mergeable. Changes to
these settings will require a reboot of the system.
Field Description
Card/Module Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks on
the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Table continues…

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 PRI Trunks

Field Description
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any
of the following features: multi-site networks, VPNremote, application telecommuter
mode.
Line Number Allocated by the system.
Line SubType Default = E1-R2
The options are:
• E1-R2
• ETSI
• QSIGA
• QSIGB
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Channel Allocation Default = 30 | 1
The order, 30 | 1 or 1 | 30, in which channels are used.
Country (Locale) Default = Mexico. Select the locale that matches the area of usage. Note that
changing the locale will return the MFC Group settings to the defaults for the selected
locale. Currently supported locales are:
• Argentina
• Brazil
• China
• India
• Korea
• Mexico
• None
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Line

Field Description
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
The table at the base of the form displays the settings for the individual channels
provided by the line. For details of the channel settings see the E1-R2 Channel form.
To edit a channel, either double-click on it or right-click and select Edit. This will
display the Edit Channel dialog box. To edit multiple channels at the same time select
the channels whilst pressing the Shift or Ctrl key. Then right-click and select Edit.

Related links
E1 R2 Line on page 352

E1-R2 Channels
Navigation: Line | E1–R2 Channels
The channel settings are split into two sub-tabs, E1R2 Edit Channel and Timers.
The Timers tab displays the various timers provided for E1-R2 channels. These should only be
adjusted when required to match the line provider's settings.
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel,
select the required channel or channels and click Edit.
The following settings are mergeable: Incoming Group ID, Outgoing Group ID, Admin.
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Descriptions
Channel The channel or channels being edited.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
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 PRI Trunks

Field Descriptions
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Direction Default = Both Directions
The direction of calls on the channel. The options are: Incoming, Outgoing, Both
Directions.
Bearer Default = Any
The type of traffic carried by the channel. The options are: Voice, Data, Any.
Admin Default = Out of Service.
This field can be used to indicate whether the channel is in use or not. On trunks where
only a limited number of channels have been requested from the trunk provider (known
as sub-equipped trunks), those channels not provided should be set as Out of Service.
For channels that are available but are temporarily not being used select Maintenance.
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Line

Field Descriptions
Line Signaling Default = R2 Loop Start
Type
The signaling type used by the channel. Current supported options are:
• R2 Loop Start
• R2 DID
• R2 DOD
• R2 DIOD
• Tie Immediate Start
• Tie Wink Start
• Tie Delay Dial
• Tie Automatic
• WAN Service
• Out of Service
Dial Type Default = MFC Dialing
The type of dialing supported by the channel. The options are: MFC Dialing, Pulse
Dialing, DTMF Dialing.

Related links
E1 R2 Line on page 352

E1 R2 MFC Group
Navigation: Line | E1–R2 MFC Group
These settings are not mergeable. Changes to these settings will require a reboot of the system.
These tabs show the parameter assigned to each signal in an MFC group. The defaults are set
according to the Country (Locale) on the Line tab. All the values can be returned to default by the
Default All button on the Advanced tab.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
To change a setting either double-click on it or right-click and select Edit.
Related links
E1 R2 Line on page 352

E1-R2 Advanced
Navigation: Line | E1R2 Advanced
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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 PRI Trunks

Field Description
Zero Suppression Default = HDB3
Selects the method of zero suppression used (HDB3 or AMI).
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used
is described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
Line Signaling Default = CPE
The options are:
• CPE
• CO
• CO
The feature is intended to be used primarily as a testing aid. It allows T1 and E1 lines
to be tested in a back-to-back configuration, using crossover (QSIG) cables.
The CO feature operates by modifying the way in which incoming calls are
disconnected for system configuration in Brazil and Argentina. In these locales, the
CO setting uses Forced-Release instead of Clear-Back to disconnect incoming calls.
The Brazilian Double-Seizure mechanism used to police Collect calls, is also disabled
in CO mode.
Incoming Routing Default = 4
Digits
Sets the number of incoming digits used for incoming call routing.
CRC Checking Default = On
Switches CRC on or off.
Default All Group Default the MFC Group tab settings.
Settings
Line Signaling Timers To edit one of these timers, either double-click on the timer or right-click on a timer
and select the action required.

Related links
E1 R2 Line on page 352

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T1 Line
Related links
PRI Trunks on page 343
US T1 Line on page 358
T1 Channels on page 360

US T1 Line
Navigation: Line | US T1 Line
The following settings are mergeable:
• Admin
• Prefix
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Line Number Allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
Line Sub Type Default = T1
Set to T1 for a T1 line.
Channel Allocation Default = 24 | 1
The order, 24 to 1 or 1 to 24, in which channels are used.
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if all users
must dial a prefix to access an outside line. The prefix is automatically placed in front of
all incoming numbers so that users can dial the number back.
Table continues…

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 PRI Trunks

Field Description
Framing Default = ESF
Selects the type of signal framing used. The options are:
• ESF
• D4
Zero Suppression Default = B8ZS
Selects the method of zero suppression used. The options are:
• B8ZS
• AMI ZCS
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from this
line. Preference should always be given to using the clock source from a central office
exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available, Fallback
can be used to specify a clock source to use should the Network source not be
available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
Haul Length Default = 0-115 feet.
Sets the line length to a specific distance.
Channel Unit Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided by the
Central Office. The options are:
• Foreign Exchange
• Special Access
• Normal
CRC Checking Default = On
Turns CRC on or off.
Line Signaling Default = CPE
This field affects T1 channels set to Loop-Start or Ground-Start. The field can be set
to either CPE (Customer Premises Equipment) or CO (Central Office). This field should
normally be left at its default of CPE. The setting CO is normally only used in lab
back-to-back testing.
Table continues…

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Field Description
Incoming Routing Default=0 (present call immediately)
Digits
Sets the number of routing digits expected on incoming calls. This allows the line to
present the call to the system once the expected digits have been received rather than
waiting for the digits timeout to expire. This field only affects T1 line channels set to E&M
Tie, E&M DID, E&M Switched 56K and Direct Inward Dial.
CSU Operation Enable this field to enable the T1 line to respond to loop-back requests from the line.
Enhanced Called Default = Off
Party Number
This option is not supported for systems set to the United States locale. Normally the
dialed number length is limited to 15 digits. Selecting this option increases the allowed
dialed number length to 30 digits.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.

Related links
T1 Line on page 358

T1 Channels
Navigation: Line | T1 Channels
The settings for each channel can be edited. Users have the option of editing individual channels
by double-clicking on the channel or selecting and editing multiple channels at the same time.
Note that the Line Appearance ID cannot be updated when editing multiple channels.
When editing a channel or channels, the settings available are displayed on two sub-tabs; T1 Edit
Channel and Timers.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Channel Allocated by the system.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Table continues…

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 PRI Trunks

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance
ID must be unique and not match any extension number. Line appearance is not
supported for trunks set to QSIG operation and is not recommended for trunks be used
for DID.
Direction Default = Bothway
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Bothway
Bearer Default = Any
The type of traffic carried by the channel. The options are: Voice, Data, Any.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Table continues…

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Field Description
Type Default = Loop-Start.
The T1 emulates the following connections:
• Ground-Start
• Loop-Start
• E&M - TIE
• E&M - DID
• E&M Switched 56K
• Direct Inward Dial
• Clear Channel 64K
Trunks set to E&M - DID will only accept incoming calls.
If E&M - TIE is selected and the Outgoing Trunk Type is set to Automatic, no
secondary dial tone is provided for outgoing calls on this line/trunk.
Dial Type Default = DTMF Dial
Select the dialing method required. The options are: DTMF Dial, Pulse Dial.
Incoming Trunk Default = Wink-Start
Type
Used for E&M types only. The handshake method for incoming calls. The options are
Outgoing Trunk Default = Wink-Start
Type
Used for E&M types only. The handshake method for outgoing calls. The options are:
Automatic, Immediate, Delay Dial, Wink-Start.
If the line Type is set to E&M-TIE and the Outgoing Trunk Type is set to Automatic, no
secondary dial tone is provided for outgoing calls on this line/trunk.
Tx Gain Default = 0dB.
The transmit gain in dB.
Rx Gain Default = 0dB.
The receive gain in dB.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.

Timer Settings
This sub-tab allows various timers relating to operation of an individual channel to be adjusted.
These should only be adjusted to match the requirements of the line provider. The following is a
list of the default values. To reset a value, click on the current value and then right click and select
from the default, minimize and maximize options displayed.

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 PRI Trunks

Incoming Automatic Delay: 410. Silent Interval: 1100.

Incoming Wink Delay: 100. Outgoing Seizure: 10.
Wink Signal: 200. Wink Start: 5000.
Incoming Dial Guard: 50. Wink Validated: 80.
First Incoming Digit: 15000. Wink End: 350.
Incoming Inter Digit: 5000. Delay End: 5000.
Maximum Inter Digit: 300. Outgoing Dial Guard: 590.
Flash Hook Detect: 240. Outgoing IMM Dial Guard: 1500.
Incoming Disconnect: 300. Outgoing Pulse Dial Break: 60.
Incoming Disconnect Guard: 800. Outgoing Pulse Dial Make: 40.
Disconnected Signal Error: 240000. Outgoing Pulse Dial Inter Digit: 720.
Outgoing Disconnect: 300. Outgoing Pulse Dial Pause: 1500.
Outgoing Disconnect Guard: 800. Flash Hook Generation: 500.
Ring Verify Duration: 220. Outgoing End of Dial: 1000.
Ring Abandon: 6300. Answer Supervision: 300.
Ping Verify: 600. Incoming Confirm: 20.
Long Ring Time: 1100.

Related links
T1 Line on page 358

T1 PRI Line
Related links
PRI Trunks on page 343
T1 ISDN on page 363
T1 ISDN Channels on page 367
T1 ISDN TNS on page 369
T1 ISDN Special on page 370
Call By Call (US PRI) on page 370

T1 ISDN
Navigation: Line | T1 ISDN Line
The following settings are mergeable:
• Prefix
• Send Redirecting Number
• Admin
• Send original calling party for forwarded and twinning calls

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• Originator number for forwarded and twinning calls

The remaining settings are not mergeable. Changes to these settings require a system reboot.
Variable Description
Line Number Allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
Line Sub Type : Default = PRI
Set to PRI. If set to T1 see Line Form (T1). If set to ETSI, ETSI CHI, QSIG A or QSIG B
see Line (E1).
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Channel Allocation Default = 23 | 1
The order, 23 to 1 or 1 to 23, in which channels are used.
Switch Type Default = NI2
The options are: 4ESS, 5ESS, DMS100, NI2.
Provider Default = Local Telco
Select the PSTN service provider (AT&T, Sprint, WorldCom or Local Telco).
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if all users
must dial a prefix to access an outside line. The prefix is automatically placed in front of
all incoming numbers so that users can dial the number back.
Table continues…

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 PRI Trunks

Variable Description
Add 'Not-end-to- Default = Never*.
end ISDN'
Sets whether the optional 'Not end-to-end ISDN' information element should be added to
Information
outgoing calls on the line. The options are: Never, Always, POTS (only if the call was
Element
originated by an analog extension).
*The default is Never except for the following locales; for Italy the default is POTS, for
New Zealand the default is Always.
Progress Default = None.
Replacement
Progress messages are defined in the Q.931 ISDN connection control signaling protocol.
Generally, if a progress message is sent, the caller does not get connected and so
typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages. The
options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not hear
the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
Send Redirecting Default = Off.
Number
This option can be used on ISDN trunks where the redirecting service is supported by
the trunk provider. Where supported, on twinned calls the caller ID of the original call is
passed through to the twinning destination. This option is only used for twinned calls.
Send Names This option is available when the Switch Type above is set to DMS100. If set, names are
sent in the display field. The Z shortcode character can be used to specify the name to
be used.
Names Length Set the allowable length for names, up to 15 characters, when Send Names is set
above.
Test Number Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
Framing Default = ESF
Selects the type of signal framing used (ESF or D4).
Zero Suppression Default = B8ZS
Selects the method of zero suppression used (B8ZS or AMI ZCS).
Table continues…

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Line

Variable Description
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from this
line. Preference should always be given to using the clock source from a central office
exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available, Fallback
can be used to specify a clock source to use should the Network source not be
available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
CSU Operation Tick this field to enable the T1 line to respond to loop-back requests from the line.
Haul Length Default = 0-115 feet
Sets the line length to a specific distance.
Channel Unit Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided by the
Central Office. The options are: Foreign Exchange, Special Access, Normal.
CRC Checking Default = On
Turns CRC on or off.
Line Signaling The field can be set to either CPE (Customer Premises Equipment) or CO (Central
Office). This field should normally be left at its default of CPE. The setting CO is normally
only used in lab back-to-back testing.
Incoming Routing Default=0 (present call immediately)
Digits
Sets the number of routing digits expected on incoming calls. This allows the line to
present the call to the system once the expected digits have been received rather than
waiting for the digits timeout to expire. This field only affects T1 line channels set to E&M
Tie, E&M DID, E&M Switched 56K and Direct Inward Dial.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Send original Default = Off.
calling party for
Use the original calling party ID when forwarding calls or routing twinned calls.
forwarded and
twinning calls This setting applies to the following ISDN lines:
• PRI24 with subtypes: PRI, QSIGA, QSIGB, ETSI, ETSI CHI.
• PRI30 with subtypes: QSIGA, QSIGB, ETSI, ETSI CHI.
Table continues…

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 PRI Trunks

Variable Description
Originator number Default = blank.
for forwarded and
The number used as the calling party ID when forwarding calls or routing twinned calls.
twinning calls
This field is grayed out when the Send original calling party for forwarded and
twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes: PRI, QSIGA, QSIGB, ETSI, ETSI CHI.
• PRI30 with subtypes: QSIGA, QSIGB, ETSI, ETSI CHI.

Related links
T1 PRI Line on page 363

T1 ISDN Channels
Navigation: Line | T1 ISDN Channels
This tab allows settings for individual channels within the trunk to be adjusted. This tab is not
available for trunks sets to ETSI or QSIG mode.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Channel Allocated by the system.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Table continues…

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Line

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance
ID must be unique and not match any extension number.
Direction Default = Both Directions
The direction of calls on the channel. The options are: Incoming, Outgoing, Both
Directions.
Bearer Default = Any
The type of traffic carried by the channel. The options are: Voice, Data, Any.
Table continues…

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 PRI Trunks

Field Description
Service Default = None.
If the line provider is set to AT&T, select the type of service provided by the channel. The
options are:
• Call by Call
• SDN (inc GSDN)
• MegaCom 800
• MegaCom
• Wats
• Accunet
• ILDS
• I800
• ETN
• Private Line
• AT&T Multiquest
For other providers, the service options are None or No Service.
Admin Default = Out of Service
Used to indicate the channel status. The options are: In Service, Out of Service,
Maintenance.
Tx Gain Default = 0dB
The transmit gain in dB
Rx Gain Default = 0dB
The receive gain in dB.

Related links
T1 PRI Line on page 363

T1 ISDN TNS
Navigation: Line | T1 ISDN TNS
This tab is shown when the line Provider is set to AT&T. It allows the entry of the Network
Selection settings. These are prefixes for alternative long distance carriers. When a number dialed
matches an entry in the table, that pattern is stripped from the number before being sent out. This
table is used to set field in the TNS (Transit Network Selection) information element for 4ESS and
5ESS exchanges. It is also used to set fields in the NSF information element.
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Line

Field Description
TNS Code The pattern for the alternate long distance carrier. For example: The pattern 10XXX is
added to this tab. If 10288 is dialed, 10 is removed and 288 is placed in the TNS and
NSF information.

Related links
T1 PRI Line on page 363

T1 ISDN Special
Navigation: Line | T1 ISDN Special
This tab is shown when the line Provider is set to AT&T. This table is used to set additional fields
in the NSF information element after initial number parsing by the TNS tab. These are used to
indicate the services required by the call. If the channel is set to Call by Call, then further parsing
is done using the records in the Call by Call tab.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Short code The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table and the Call-by-call table to
the number dialed by the user.
Number The number to be dialed to line.
Special Default = No Operator.
The options are: No Operator, Local Operator or Presubscribed Operator.
Plan Default = National.
The options are: National or International.

Typical values are:

Short Code Number Service
011N N No Operator, International
010N N Local Operator, International
01N N Local Operator, National
00N N Presubscribed Operator, National
0N N Presubscribed Operator, National
1N 1N No operator, National

Related links
T1 PRI Line on page 363

Call By Call (US PRI)

Navigation: Line | T1 ISDN Call by Call

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 PRI Trunks

This tab is shown when the line Provider is set to AT&T. Settings in this tab are only used when
calls are routed via a channel which has its Service set to Call by Call.
It allows short codes to be created to route calls to a different services according to the number
dialed. Call By Call reduces the costs and maximizes the use of facilities. Call By Call chooses the
optimal service for a particular call by including the Bearer capability in the routing decision. This is
particularly useful when there are limited resources.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Short Code The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table to the number dialed by the
user.
Number The number to be dialed to line.
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Service Default = AT&T
The service required by the call. The options are:
• Call by Call
• SDN (inc GSDN)
• MegaCom 800
• MegaCom
• Wats
• Accunet
• ILDS
• I800
• ETN
• Private Line
• AT&T Multiquest

Related links
T1 PRI Line on page 363

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Line

SIP Line
IP Office supports SIP voice calls through the addition of SIP lines to the system configuration.
This approach allows users with non-SIP phones to make and receive SIP calls.
Deleting a SIP line requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Related links
Line on page 290
SIP Line on page 372
Transport on page 376
Call Details on page 379
VoIP on page 386
T38 Fax on page 390
SIP Credentials on page 391
SIP Advanced on page 392
Engineering on page 398

SIP Line
Navigation: Line | SIP Line | SIP Line
Configuration Settings
These settings are mergeable with the exception of the Line Number setting. Changing the Line
Number setting requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/1 to 349 (Server Edition).
The line number must be unique for each line in the configuration. IP500 V2 systems
reserved line numbers 1 to 16 for internal hardware.
Table continues…

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 SIP Line

Field Description
ITSP Domain Name Default = Blank.
This field is used to specify the default host part of the SIP URI in the From, To, and
R-URI fields for outgoing calls. For example, in the SIP URI [email protected], the
host part of the URI is example.com. When empty, the default host is provided by the
SIP Line | SIP Transport | ITSP Proxy Address field value. If multiple addresses are
defined in the ITSP Proxy Address field, then this field must be defined.
For the user making the call, the user part of the From SIP URI is determined by the
settings of the SIP URI channel record being used to route the call (see SIP Line | SIP
URI | Local URI). This will use one of the following:
• a specific name entered in Local URI field of the channel record.
• or specify using the primary or secondary authentication name set for the line below.
• or specify using the SIP Name set for the user making the call (User | SIP | SIP
Name).
For the destination of the call, the user part of the To and R-URI fields are determined
by dial short codes of the form 9N/N"@example.com” where N is the user part of the SIP
URI and "@example.com" is optional and can be used to override the host part of the To
and R-URI.
Local Domain Default = Blank.
Name
An IP address or SIP domain name as required by the service provider. When
configured, the Local Domain Name value is used in
• the From and Contact headers
• the PAI header, if Line > SIP Advanced is checked
• the Diversion header
If both the ITSP Domain Name and Local Domain Name are configured, Local
Domain takes precedence.
Local Domain Name is not used in the Remote Party ID header.
URI Type Default = SIP URI.
Set the format the IP Office uses for SIP URI entries in headers.
• SIP URI - Use SIP URI format. For example, display <sip:content@hostname>
• Tel - Use Tel URI format. For example, +1-425-555-4567. This affects the From field
of outgoing calls. The To field for outgoing calls uses the format specified by the short
codes used for outgoing call routing.
• SIPS - Use SIPS format for all URIs. SIPS can be used only when Layer 4 Protocol is
set to TLS.
Table continues…

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Line

Field Description
Location Default = Cloud.
You can set Location values for the IP Office system and for individual extensions and
lines. Associating a line with a location:
• Applies the location's call admission control (CAC) settings to the line. See Configuring
Call Admission Control on page 709.
• For SIP lines that support RFC4119/RFC5139, emergency calls using the line can
include the location's address information.
• For more information, see Using Locations on page 617.
Prefix The IP Office uses these values to adjust incoming numbers to match the format required
for outgoing calls and used in system directory entries.
National Prefix
1. If the number starts with a + symbol, the symbol is replaced with the International
International Prefix
Prefix.
Country Code
2. If the Country Code has been set:
a. If the number begins with the Country Code, or International Prefix plus
Country Code, the IP Office replaces them with the National Prefix.
b. If the number does not start with the National Prefix or International Prefix,
the IP Office adds the International Prefix.
3. If the incoming number does not begin with the National Prefix or International
Prefix, the IP Office adds the Prefix.
For more details, see SIP Prefix Operation on page 851.
Name Priority Default = System Default.
For SIP trunks, the caller name displayed on an extension can either be that supplied
by the trunk or one obtained by checking for a number match in the extension user's
personal directory and the system directory. This setting determines which method is
used by the line. The options are:
• System Default: Use the system setting System | Telephony | Telephony | Default
Name Priority.
• Favor Trunk: Display the name provided by the trunk. For example, the trunk may be
configured to provide the calling number or the name of the caller. The system should
display the caller information as it is provided by the trunk. If the trunk does not provide
a name, the system uses the Favor Directory method.
• Favor Directory: Search for a number match in the extension user's personal directory
and then in the system directory. The first match is used and overrides the name
provided by the SIP line. If no match is found, the name provided by the line, if any, is
used.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The description is
not used elsewhere.
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Field Description
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows you to configure trunks as either Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using any of the
following features: multi-site networks, VPNremote, application telecommuter mode.
In Service Default = On.
When this field is not selected, the SIP trunk is unregistered and not available to
incoming and outgoing calls.
Check OOS Default = On.
If enabled, the system will regularly check if the trunk is in service using the methods
listed below. Checking that SIP trunks are in service ensures that outgoing call routing is
not delayed waiting for response on a SIP trunk that is not currently usable.
For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an
OPTIONS message is received the trunk is taken out of service.
For trunks using DNS, if the IP address is not resolved or the DNS resolution has
expired, the trunk is taken out of service.

Session Timers
Field Description
Refresh Method Default = Auto.
The options are: Auto, Reinvite or Update.
When Auto is selected, if UPDATE is in the Allow: header from the far SIP endpoint, then
it is used. Otherwise INVITE is used.
Timer (seconds) Default = On Demand. Range = 90 to 64800
This field specifies the session expiry time. At the half way point of the expiry time, a
session refresh message is sent. When set to On Demand, IP Office will not send a
session refresh message but will respond to them.

Redirect and Transfer

Redirection and blind transfer are configured separately. By default, they are disabled.
A supervised transfer occurs when a consultation call is made and the REFER contains a
Replaces: header indicating the CallID of another call leg which the REFERing agent has already
initiated with the REFER target.
Note:
• Do not change these settings unless directed to by the SIP service provider.

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Field Description
Incoming Default = Auto.
Supervised REFER
Determines if IP Office will accept a REFER being sent by the far end. The options are:
• Always: Always accepted.
• Auto: If the far end does not advertise REFER support in the Allow: header of the
OPTIONS responses, then IP Office will reject a REFER from that endpoint.
• Never: Never accepted.
Outgoing Default = Auto.
Supervised REFER
Determines if IP Office will attempt to use the REFER mechanism to transfer a party
to a call leg which IP Office has already initiated so that it can include the CallID in a
Replaces: header. The options are:
• Always: Always use REFER.
• Auto: Use the Allow: header of the OPTIONS response to determine if the endpoint
supports REFER.
• Never: Never use REFER.
Send 302 Moved Default = Off.
Temporarily
A SIP response code used for redirecting an unanswered incoming call. It is a response
to the INVITE, and cannot be used after the 200 OK has been sent as a response to the
INVITE.
Outgoing Blind Default = Off.
REFER
When enabled, a user, voicemail system or IVR can transfer a call by sending a REFER
to an endpoint that has not set up a second call. In this case, there is no Replaces:
header because there is no CallID to replace the current one. This directs the far end to
perform the transfer by initiating the new call and release the current call with IP Office.

Related links
SIP Line on page 372

Transport
Navigation: Line | SIP Line | Transport
Behavior during Service unavailable
A proxy server is considered Active once the system has received a response to an INVITE,
REGISTER or OPTIONS.
In the case of the proxy server responding with 503 - Service Unavailable, it should be
considered Active - In Maintenance. In this case, the following should occur:
• If the response 503 - Service Unavailable was in response to an INVITE request:
- If calls are tied to registrations (Calls Route via Registrar enabled) and there are
other proxies available, the tied registrations should issue an Un-REGISTER and try to
REGISTER with a different proxy. The call should fail with cause = Temporary Fail.

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- If calls are not tied, the INVITE should be immediately tried to a different proxy.
• If the response 503 - Service Unavailable was in response to a REGISTER request:
- If there are other proxies available, this registration only should issue an Un-REGISTER
and try to REGISTER with a different proxy.
- If Explicit DNS Server(s) are configured, a DNS request should be sent out to see
whether the proxy server has disappeared from those being offered.
An Active-InMaintenance proxy server should not be used for a new transactions (INVITE or
REGISTER) until:
• There is a change in DNS responses indicating the proxy has become active.
• The configuration does not leave any better option available. In this case, there should be a
throttle so that no more than 5 failures (without successes) in 1 minute should be allowed.
• A configuration merge has occurred where the ITSP Proxy Address has been changed.
• 10 minutes has expired.
Behavior during Not Responding
A proxy server that is not-responding (UDP) is indicated when 3 requests are sent and no replies
are received. This would normally occur during a single INVITE transaction.
Consideration should be given whether this is caused by a local network fault or is caused by the
Proxy being out of service. Since it is likely to be local, no action should be taken unless traffic is
received from an alternative proxy while this proxy is actually not responding. The state should be
"Possibly non responding".
If explicit DNS servers are configured, a DNS request should be sent out to see whether this
Proxy server has disappeared from those being offered.
If possible, an alternative proxy should be stimulated simultaneously with stimulating the suspect
server.
The server should be considered non-responding if it is persistently non-responding while
other proxies are responding or if it is non-responding and has disappeared from the DNS
advertisement.
While in the "possibly not responding" state, it would be better to send an INVITE to an alternative
proxy while simultaneously sending any appropriate message to this proxy. This will help to
resolve whether it is really not responding rather than there being local network problems.
However, there is no requirement to blacklist the proxy.
Once in the "definitely not responding" state:
• If there are other proxies available: this registration only issues an Un-REGISTER, and try to
REGISTER with a different proxy. Calls do not automatically clear.
• If a SIP message is received from it, the state should immediately go"Active".
• This proxy should be blacklisted unless there are no better options available. While
blacklisted, only one transaction per 10 minutes is allowed.
• Even if not blacklisted, there should be a throttle so that no more than 5 failures (without
successes) in 1 minute should be allowed.

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Configuration settings
The ITSP Proxy Address and Calls Route via Registrar settings are mergeable. Changing the
remaining settings requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
Field Description
ITSP Proxy Address Default = Blank
This is the SIP Proxy address used for outgoing SIP calls. The address can be
specified in the following ways:
• If left blank, the ITSP Domain Name is used and is resolved by DNS resolution in
the same way as if a DNS address had been specified as below.
• An IP address.
• A list of up to 4 IP addresses, with each address separated by a comma or space.
- The addresses can include an indication of the relative call weighting of each
address compared to the others. This is done by adding a w N suffix to the
address where N is the weighting value. For example, in the list 213.74.81.102w3
213.74.81.100w2, the weighting values assigns 1.5 times the weight of calls to the
first address. The default weight if not specified is 1. A weight of 0 can be used to
disable an address. Weight is only applied to outgoing calls.
If there is more than one proxy defined, and no weight indication, then calls are
only sent to the first in the list until there is a failure at which point the next proxy is
used.
- If the Calls Route via Registrar setting below is enabled, the weighting is applied
to registrations rather than calls.
• A DNS address, for example sbc.example.com.
- The DNS response may return multiple proxy addresses (RFC 3263). If that is
the case, the system will resolve the address to use based on priority, TTL and
weighting information included with each address.
- A load balancing suffix can be added to specify that multiple proxy results should
be returned if possible, for example sbc.example.com(N). where N is the required
number of addresses from 1 to 4.
This field is mergeable. However, no more than 4 IP Addresses should be in use
at any time. So, if the combined new and old address settings exceed 4, the new
addresses are only phased into use as transactions in progress on the previous
addresses are completed.
Network Configuration
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Field Description
Layer 4 Protocol Default = UDP.
The options are: TCP, UDP or TLS.
• TLS connections support the
following ciphers: TLS_RSA_WITH_AES_128_CBC_SHA,
TLS_RSA_WITH_AES_256_CBC_SHA, TLS_DHE_RSA_WITH_AES_128_CBC_SHA,
and TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Use Network Default = None.
Topology Info
• LAN1 - Associate the line with the Network Topology and DiffServ Settings
settings of IP Office LAN1.
- If no STUN server address is set for the LAN interface, then the Binding Refresh
Time is ignored when calculating the timing for periodic OPTIONS messages
unless the Firewall/NAT Type is set to Open Internet.
• LAN2 - As above but using the settings of IP Office LAN2.
• None - If selected, the IP Office does not apply STUN lookup. The IP Office system
IP routing tables determine routing for the line.
Send Port When the Layer 4 Protocol is set to TLS, the default port is 5061. When set to TCP
or UDP, the default port is 5060.
Listen Port When the Layer 4 Protocol is set to TLS, the default port is 5061. When set to TCP
or UDP, the default port is 5060.
Explicit DNS Default = 0.0.0.0 (Off)
Server(s)
If specific DNS servers should be used for SIP trunk operation rather than the
general DNS server specified or obtained for the system, the server addresses can be
specified here. If exported or imported as part of a trunk template.
Calls Route via Default = On
Registrar
If selected, all calls are routed via the same proxy as used for registration. If multiple
ITSP proxy addresses have been specified, there is no load balancing of registrations.
Separate Registrar Default = Blank
This field allows the SIP registrar address to be specified if it is different from that of
the SIP proxy. The address can be specified as an IP address or DNS name.

Related links
SIP Line on page 372

Call Details
Navigation: Line | SIP Line | Call Details
These settings are used to control the incoming and outgoing calls that use the SIP line. They also
set the SIP headers used on calls and the source for values within those headers.

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Description
SIP URIs These settings are used for general incoming and outgoing calls on the SIP line.
SIP Line These settings allow the emulation of line appearance operation by the SIP line.
Appearances

For details of how these are used as part of call routing, see Outgoing SIP Call Routing on
page 841.
Related links
SIP Line on page 372
SIP URIs on page 380
SIP Line Appearances on page 383

SIP URIs
For the IP Office, each SIP URI acts as a set of trunk channels. It also sets the content of various
SIP headers and how that content is used.
• For outgoing calls, the IP Office maps internal calling or called numbers to headers to match
the ITSPs requirements. Outgoing calls are routed to a SIP URI by short codes that match
the URIs Outgoing Group setting. See SIP Outgoing Call Routing on page 841.
• For incoming calls, headers in the SIP message are used for call routing. Incoming calls
are routed to incoming call routes that match the URI's Incoming Group setting. See SIP
Incoming Call Routing on page 849.
• The IP Office supports up to 150 SIP URIs on each SIP line.
General Settings
Name Description
URI This field is for information only and cannot be edited.
Incoming Group Default = 0, Range 0 to 99999.
This value is used to match incoming to the Line Group ID of an incoming call route
entry. See SIP Incoming Call Routing on page 849.
Outgoing Group Default = 0, Range 0 to 99999.
Short codes that specify a number to dial to a line specify a Line Group ID. This is
used to match to lines with the same Outgoing Group value. See SIP Outgoing Call
Routing on page 841.
Max Sessions Default =10
This field sets the maximum number of simultaneous calls that can use the URI before
the system returns busy to any further calls.
Credentials Default = 0:<None>
This field is used to select from a list of the account credentials configured on the line's
SIP Credentials tab.

The remaining sections are arranged as a table of values. These set which SIP headers are used
for calls routed by the SIP URI entry.

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The table also sets the source of the values used in the SIP URI values in those headers. A typical
SIP URI takes the following form: display <sip:content@hostname> where:
• display is the displayed name value for the caller/called party.
• content is the call target name or number.
• hostname is the host from/to which the calls are sent. For details of how the hostname used
by the IP Office system is set. See Setting the SIP URI Host on page 837.
Headers
The first column indicates the headers used for calls matched to this SIP URI entry.
Name Description
Local URI Default = Auto
This field sets the From field for outgoing SIP calls using this URI.
Contact Default = Auto
This field sets the From field for outgoing SIP calls using this URI.
P Asserted ID Default = Disabled
When selected, identity information is provided in P-Asserted-Identity (PAI)
headers.
P Preferred ID Default = Disabled
When selected, identity information is provided in a P-Preferred-Identity header.
Diversion Header Default = Disabled
When selected, information from the Diversion Header is provided in the SIP
messages.
Remote Party ID Default = Disabled
When selected, Remote Party ID header are provided with calls.

Display
This column sets the source for the display part of the SIP URI used in the selected headers.
Setting Description
Auto If Auto is selected, the system automatically determines the appropriate value to use. It
uses external numbers when forwarding incoming calls, and internal extension numbers
for calls made by a local user.
• On incoming calls, the system looks for matches against extension numbers and
system short codes.
• On outgoing calls, the system allows short code manipulation of the caller number and
name. For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
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Setting Description
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead. See Anonymous SIP Calls on
page 842.
Manual Entry If required, you can manually type in a value to use. The value is then used by other
fields configured as Explicit. This is typically used to set the DDI to be associated with
SIP line appearances.
Credential Values If a Credentials entry has been selected above, then the User name, Authentication
Name and Contact values from the selected credentials entry can be selected as values.
The value is then used by other fields configured as Explicit.
• URI values should only be set using credentials when required by the line provider.
For example, some line providers require the From header to always contains the
credentials used for registration, whilst other headers are used to convey information
about the caller ID.

Content
This column sets the source for the content part of the SIP URI used in the selected headers.
Setting Description
Auto If Auto is selected, the system automatically determines the appropriate value to use. It
uses external numbers when forwarding incoming calls, and internal extension numbers
for calls made by a local user.
• On incoming calls, the system looks for matches against extension numbers and
system short codes.
• On outgoing calls, the system allows short code manipulation of the caller number and
name. For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead. See Anonymous SIP Calls on
page 842.
Manual Entry If required, you can manually type in a value to use. The value is then used by other
fields configured as Explicit. This is typically used to set the DDI to be associated with
SIP line appearances.
Credential Values If a Credentials entry has been selected above, then the User name, Authentication
Name and Contact values from the selected credentials entry can be selected as values.
The value is then used by other fields configured as Explicit.
• URI values should only be set using credentials when required by the line provider.
For example, some line providers require the From header to always contains the
credentials used for registration, whilst other headers are used to convey information
about the caller ID.

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Field Meaning
These values are used to set the source or value for headers based on the call direction.
Field Description
Outgoing Calls Set the source for URI header information on outgoing external calls.
Forwarding/ Set the source for URI header information on calls being forwarded externally.
Twinning
Incoming Calls Set the source for URI header information on incoming external calls.

The following values can be selected for the different fields.

Field Description
Caller Use the values associated with the calling party. For forwarded calls, use the values
associated with the party forwarding the call.
Original Caller For forwarded calls, use the value associated with the original caller.
Called Use the values associated with the called party.
Explicit Use the manual entered values from the header Display and Content fields, or the
credentials values selected from those drop-downs.
None Do not send the header.

Related links
Call Details on page 379

SIP Line Appearances

These settings allow the SIP line to emulate the use of line appearances on phones that support
line appearance buttons. Those buttons can then be used to make or receive calls. For details,
see SIP Line Appearances on page 864.
SIP line appearances are not supported over a multi-site network/SCN or in resiliency.
General Settings
Name Description
SIP Line Default = Disabled
Appearances
When enabled, SIP line appearances can be configured. Note, if you disable this setting
and save the configuration, all configured SIP line appearance values are removed.
Incoming Group Default = 0, Range 0 to 99999.
This value is used to match incoming to the Line Group ID of an incoming call route
entry. See SIP Incoming Call Routing on page 849.
Outgoing Group Default = 0, Range 0 to 99999.
Short codes that specify a number to dial to a line specify a Line Group ID. This is
used to match to lines with the same Outgoing Group value. See SIP Outgoing Call
Routing on page 841.
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Name Description
Credentials Default = 0:<None>
This field is used to select from a list of the account credentials configured on the line's
SIP Credentials tab.
Max Sessions Default =10
This field sets the maximum number of simultaneous calls that can use the URI before
the system returns busy to any further calls.
Incoming Sessions Default = 3
The maximum number of incoming call sessions.
Outgoing Sessions Default = 3
The maximum number of outgoing call sessions. .

The remaining sections are arranged as a table of values. These set which SIP headers are used
for calls routed by the SIP URI entry.
The table also sets the source of the values used in the SIP URI values in those headers. A typical
SIP URI takes the following form: display <sip:content@hostname> where:
• display is the displayed name value for the caller/called party.
• content is the call target name or number.
• hostname is the host from/to which the calls are sent. For details of how the hostname used
by the IP Office system is set, see Setting the SIP URI Host on page 837.
Headers
The first column indicates the headers used for calls matched to this SIP URI entry.
Name Description
Local URI Default = Auto
This field sets the From field for outgoing SIP calls using this URI.
Contact Default = Auto
This field sets the From field for outgoing SIP calls using this URI.
P Asserted ID Default = Disabled
When selected, identity information is provided in P-Asserted-Identity (PAI)
headers.
P Preferred ID Default = Disabled
When selected, identity information is provided in a P-Preferred-Identity header.
Diversion Header Default = Disabled
When selected, information from the Diversion Header is provided in the SIP
messages.
Remote Party ID Default = Disabled
When selected, Remote Party ID header are provided with calls.

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Display
This column sets the source for the display part of the SIP URI used in the selected headers.
Setting Description
Auto If Auto is selected, the system automatically determines the appropriate value to use. It
uses external numbers when forwarding incoming calls, and internal extension numbers
for calls made by a local user.
• On incoming calls, the system looks for matches against extension numbers and
system short codes.
• On outgoing calls, the system allows short code manipulation of the caller number and
name. For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead. See Anonymous SIP Calls on
page 842.
Manual Entry If required, you can manually type in a value to use. The value is then used by other
fields configured as Explicit. This is typically used to set the DDI to be associated with
SIP line appearances.
Credential Values If a Credentials entry has been selected above, then the User name, Authentication
Name and Contact values from the selected credentials entry can be selected as values.
The value is then used by other fields configured as Explicit.
• URI values should only be set using credentials when required by the line provider.
For example, some line providers require the From header to always contains the
credentials used for registration, whilst other headers are used to convey information
about the caller ID.

Content
This column sets the source for the content part of the SIP URI used in the selected headers.
Setting Description
Auto If Auto is selected, the system automatically determines the appropriate value to use. It
uses external numbers when forwarding incoming calls, and internal extension numbers
for calls made by a local user.
• On incoming calls, the system looks for matches against extension numbers and
system short codes.
• On outgoing calls, the system allows short code manipulation of the caller number and
name. For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
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Setting Description
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead. See Anonymous SIP Calls on
page 842.
Manual Entry If required, you can manually type in a value to use. The value is then used by other
fields configured as Explicit. This is typically used to set the DDI to be associated with
SIP line appearances.
Credential Values If a Credentials entry has been selected above, then the User name, Authentication
Name and Contact values from the selected credentials entry can be selected as values.
The value is then used by other fields configured as Explicit.
• URI values should only be set using credentials when required by the line provider.
For example, some line providers require the From header to always contains the
credentials used for registration, whilst other headers are used to convey information
about the caller ID.

Field Meaning
These values are used to set the source or value for headers based on the call direction.
Field Description
Outgoing Calls Set the source for URI header information on outgoing external calls.
Incoming Calls Set the source for URI header information on incoming external calls.

The following values can be selected for the different fields.

Field Description
Explicit Use the manual entered values from the header Display and Content fields, or the
credentials values selected from those drop-downs.
None Do not send the header.

Related links
Call Details on page 379

VoIP
Navigation: Line | SIP Line | VoIP
This form is used to configure the VoIP settings applied to calls on the SIP trunk.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the order
of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
Fax Transport Default = None.
Support
This option is available only if Re-Invite Supported is selected.
• IP500 V2 systems can terminate T38 fax calls.
• Linux-based IP Office systems can route the calls between trunks/terminals with
compatible fax types.
• Set the method the IP Office uses to handle fax calls.
The supported options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does not
support T38, the IP Office will send a re-invite to change the transport method to
G.711.
DTMF Support Default = RFC2833 (IP500 V2), RFC2833/RFC4733 (Linux-Based Server)
Selects the method the IP Office uses to signal DTMF key press digits to the remote end.
The options are:
• In Band - Send digits as part of the audio path.
• RFC2833 or RFC2833/RF4733 - Send digits using a separate audio stream from the
voice path. If not supported by the far end, the line reverts to using In Band signaling.
• Info - Send the digits in SIP INFO packets.
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Field Description
Media Security Default = Disabled.
These setting control whether SRTP is used for this line and the settings used for the
SRTP. The options are:
• Same as System: Matches the system setting at System | System | VoIP | VoIP
Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data) is
enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on System
| System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be protected
using encryption. The default is to encrypt just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be protected
using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any audio
data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at both
ends.
Local Hold Music Default = Off.
When enabled, if the far end puts the call on HOLD, the system plays music received
from far end (SIP Line) to the other end. RTCP reports are sent towards SIP Line. When
disabled, the system plays local music to the other endpoint and no RTCP packets are
sent to SIP trunk.
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Field Description
Re-Invite Default = Off.
Supported
When enabled, the IP Office can use Re-Invite during a call to change the
characteristics of the call. For example, when the target of an incoming call or a transfer
does not support the codec originally negotiated on the trunk.
• Requires the ITSP to also support Re-Invite.
• This setting must be enabled for video support.
Codec Lockdown Default = Off.
In response to a SIP offer with a list of codecs, some SIP user agents send a SDP
answer that also lists multiple codecs. The user agent can then switch to any of those
codecs during the session without requiring further negotiation. However, IP Office does
not support this, so loss of speech path occurs if the current codec changes without
renegotiation.
• If enabled, when the IP Office receives an SDP answer with multiple codecs from its list
of offered codecs, the IP Office sends a re-INVITE using just a single codec from the
list, and an SIP offer with just the single chosen codec.
• This option requires Re-Invite Supported enabled.
Allow Direct Media Default = On
Path
This settings controls whether calls between IP endpoints and/or lines must go through
the IP Office or can try to route directly if possible within the customer network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a voice
compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both ends
must support direct media and have matching VoIP settings, for example using the
same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise,
the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.
PRACK/100rel Default = Off.
Supported
When selected, supports Provisional Reliable Acknowledgment (PRACK) on SIP trunks.
Enable this parameter when you want to ensure that provisional responses, such
as announcement messages, have been delivered. Provisional responses provide
information on the progress of the request that is in process. For example, while a cell
phone call is being connected, there may be a delay while the cell phone is located; an
announcement such as “please wait while we attempt to reach the subscriber” provides
provisional information to the caller while the request is in process. PRACK, which is
defined in RFC 3262, provides a mechanism to ensure the delivery of these provisional
responses.
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Field Description
Force direct media Default = On
with phones
When enabled, if an Avaya IP phone dials digits during a direct media call, the IP Office
changes the call to indirect media and sends the digits as RFC2833. 15-seconds after
the last digit, the IP Office changes the call back to direct media.
• This setting is requires the line to have Re-Invite Supported and Allow Direct Media
Path enabled, and DTMF Support set to RFC2833/RF4733.
G.711 Fax ECAN Default = Off
When enabled, if the IP Office detects a fax call, it switches to G.711 with echo
cancellation (ECAN) based on the 'G.711 Fax ECAN field, NLP disabled, a fixed jitter
buffer, and silence suppression is disabled. You can use this to avoid an ECAN mismatch
with the trunk provider.
• This setting is only available on IP500 V2 systems when Fax Transport Support is set
to G.711 or T38 Fallback.

Related links
SIP Line on page 372

T38 Fax
Navigation: Line | SIP Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which they
both support. The options are: 0, 1, 2, 3.
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For UDPTL,
redundancy error correction is supported. Forward Error Correction (FEC) is not
supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed V.21
T.30 fax transmissions.
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 SIP Line

Field Description
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27 and
V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax equipment or is
found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Default = 150.
Timeout (secs)
Scan Line Fix-up Default = On.
TFOP Default = On.
Enhancement
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression
NSF Override Default = Off.
If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be
overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
SIP Line on page 372

SIP Credentials
Navigation: Line | SIP Line | SIP Credentials
These settings in the SIP Credentials tab are used to enter the ITSP username and password
for the SIP account with the ITSP. If you have several SIP accounts going to the same ITSP IP
address or domain name, you can enter up to 30 sets of ITSP account names and passwords on
this tab.
Use the Add, Remove, and Edit buttons to manage the set of credentials for the SIP trunk
accounts.

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Line

Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Descriptions
Index This number is assigned automatically and cannot be edited. If the From field on the
SIP URI being used for the call is set to Use Authentication Name , the registration
field of the SIP URI indicates the index number of the SIP credentials to use for calls
by that SIP URI.
User Name This name must be unique and is used to identify the trunk. The name can include the
domain if necessary.
Authentication Name Default = Blank.
This field can be blank but must be completed if a Password is also specified. This
value is provided by the SIP ITSP. Depending on the settings on the Local URI tab
associated with the SIP call, it may also be used as the user part of the SIP URI. The
name can include the domain if necessary.
Contact Default = Blank.
This field is used to enter a contact and can include the domain if necessary.
Password Default = Blank.
This value is provided by the SIP ITSP. If a password is specified, the matching
Authentication Name must also be set.
Expiration (mins) Default = 60 minutes.
This setting defines how often registration with the SIP ITSP is required following any
previous registration.
Registration Default = On.
Required
If selected, the fields above above are used for registration when making calls. If
exported or imported as part of a trunk template.

Related links
SIP Line on page 372

SIP Advanced
Navigation: Line | SIP Line | SIP Advanced
Additional configuration information
For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 621.
Configuration settings
These settings are mergeable, with the exception of the Media Connection Preservation setting.
• Changing the Media Connection Preservation setting requires a “merge with service
disruption”. When the configuration file is sent to the system, the SIP trunk is restarted and all
calls on the line are dropped.

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 SIP Line

Association Method
When the IP Office receives an incoming SIP call, it needs to match the call to one of its SIP line.
• Lines are checked for a match in Line Number order until a match occurs.
• The method used to check for a match on a line uses the line's Association Method.
• If no match occurs on any line, the request is ignored.
This process enables support of multiple SIP lines with the same address settings. For example,
for scenarios that require support of multiple SIP lines from the same ITSP. That can occur when
the same ITSP supports different call plans on separate lines, or where all outgoing SIP lines are
routed from the system through an additional on-site system.
Field Description
By Source IP Address Uses the source IP address and port of the incoming request for association. The
match is against the configured remote end of the SIP line, using either an IP address/
port or resolution of an FQDN. For UDP calls, the local Listen Port is also used for
the match.
• For TCP/TLS connections, the IP Office establishes a connection to the remote
address and port specified on the SIP line.
• For UDP, non-call dialogs and call starting dialogs must use the remote address and
port specified on the SIP line.
It is recommended that the remote end does not change these value as that may
prevent NAT traversal.
"From" header Uses the host part of the From header in the incoming SIP request for association.
hostpart against ITSP
• The match is against the Line > SIP Line > ITSP Domain Name.
domain
R-URI hostpart Uses the host part of the Request-URI header in the incoming SIP request for
against ITSP domain association.
• The match is against the Line > SIP Line > ITSP Domain Name.
"To" header hostpart Uses the host part of the To header in the incoming SIP request for association.
against ITSP domain
• The match is against the Line > SIP Line > ITSP Domain Name.
"From" header Uses the host part of the From header in the incoming SIP request for association.
hostpart against
• The match is found by comparing the From header against the IP address
DNS-resolved ITSP
domain resolution of the Line > SIP Line > ITSP Domain Name or, if set, the Line > SIP
Transport > ITSP Proxy Address setting.
"Via" header hostpart Uses the host part of the VIA header in the incoming SIP request for association.
against DNS-resolved
• The match is found by comparing the VIA header against the IP address resolution
ITSP domain
of the Line > SIP Line > ITSP Domain Name or, if set, the Line > SIP Transport >
ITSP Proxy Address setting.
"From" header Uses the host part of the From header in the incoming SIP request for association.
hostpart against ITSP
• The match is against the Line > SIP Transport > ITSP Proxy Address setting.
proxy
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Line

Field Description
"To" header hostpart Uses the host part of the From header in the incoming SIP request for association.
against ITSP proxy
• The match is against the Line > SIP Transport > ITSP Proxy Address setting.
R-URI hostpart Uses the host part of the Request-URI in the incoming SIP request for association.
against ITSP proxy
• The match is against the Line > SIP Transport > ITSP Proxy Address setting.

Addressing
Field Description
Call Routing Method Default = Request URI.
This field selects which incoming SIP information is used for incoming number
matching by the IP Office to route incoming calls. The options are to match the
Request URI or the To Header element provided with the incoming call.
Use P-Called-Party Default = Off.
When enabled, IP Office reads the P-Called-Party ID header if present in the SIP
message and routes the incoming SIP calls based on it. The feature can be enabled
on public SIP trunk interfaces.
If enabled and the header is not present in the SIP message, the IP Office uses the
header configured in the Call Routing Method for incoming call routing.
Suppress DNS SRV Default = Off.
Lookups
Controls whether to send SRV queries for this endpoint, or just NAPTR and A record
queries.

Identity
Field Description
Use Phone Context Default = Off.
When enabled, signals SIP enabled PBXs that the call routing identifier is a telephone
number.
Add user=phone Default = Off.
This setting is available when Use Phone Context is enabled.
When enabled, this setting adds the SIP parameter User with value Phone to the
From and To SIP headers of outgoing calls.
Use + for Default = Off.
International
When enabled, outgoing international calls use E.164/International format with +
followed by the country code and then the telephone number.
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 SIP Line

Field Description
Use PAI for Privacy Default = Off.
When enabled, if the caller ID is withheld:
• The SIP message From header is made anonymous
• The caller identity is inserted into the P-Asserted-Identity header.
This should only be used in a trusted network and must be stripped out of the SIP
message before it is forwarded outside the trusted domain.
Use Domain for PAI Default = Off.
• When disabled, the DNS resolved IP address of the ITSP Proxy is used for the host
part in the P-Asserted-Identity header.
• When enabled, the Domain is used.
Caller ID FROM Default = Off.
Header
Incoming calls can include caller ID information in both the From field and in the PAI
fields. When this option is enabled, the caller ID information in the From field is used
rather than that in the PAI fields.
Send From In Clear Default = Off.
When enabled, the user ID of the caller is included in the From field. This applies
even if the caller has selected to be or is configured to be anonymous. However, their
anonymous state is still honored in other fields used to display the caller identity.
Cache Auth Default = On.
Credentials
When enabled, credentials challenge and response information from a registration
transaction is cached by the IP Office and automatically inserted into later SIP
messages without waiting for a subsequent challenge. This speeds up connections
but must be supported by the other end of the connection.
Add UUI header Default = Off.
When enabled, the User-to-User Information (UUI) is passed in SIP headers to
applications.
Add UUI header to Default = Off.
redirected calls
When enabled, the UUI is passed in SIP headers for calls that are redirected. For
example, on forwarded and twinned calls.
This field can be enabled if Add UUI header is enabled.
User-Agent and Default = Blank (Use system type and software level).
Server Headers
The value set in this field is used as the User-Agent and Server value included in SIP
request headers made the line.
• If blank, the type of IP Office system and its software level are used.
• Setting a unique value can be useful in call diagnostics when the IP Office has
multiple SIP trunks.
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Line

Field Description
Send Location Info Default = Never.
This option is useable with SIP ISPs that support RFC 4119/RFC 5139. When
enabled, emergency calls send the address information associated with the dialing
extension's location. See Configuration for Emergency Calls on page 652.
The options are:
• Never: Do not send location information.
• Emergency Calls: For Dial Emergency calls, send the address information
configured for the dialing extension's location.

Calling Number Verification

These settings configure the SIP trunks use of STIR protocols for calling number verification.
For more details, see SIP Calling Number Verification (STIR/SHAKEN) on page 866.
Field Description
Calling Number Default = Off
Verification
Sets whether the line uses calling number verification.
Incoming Calls Handling Default = Allow Not Failed
Set which calls are accepted by the system based on the attestation level of the
call.
• System - Use the default system setting (System VoIP > VoIP Security >
Callng Number Verification).
• Allow All - Allow all calls regardless of calling number verification.
• Allow Validated - Only accept verified calls with full or partial attestation.
• Allow Not Failed - Accept all calls expect those that specifically failed
verification. Note this can include calls with no reported verification result.

Media
Field Description
Allow Empty INVITE Default = Off.
When set to On, allows 3pcc devices to initiate calls to IP Office by sending an INVITE
without SDP.
Send Empty re- Default = Off.
INVITE
This option is only available if Line | SIP Line | VoIP | Re-Invite Supported is
selected.
If set to On, when connecting a call between two endpoints, IP Office sends an
INVITE without SDP in order to solicit the full media capabilities of both parties.
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 SIP Line

Field Description
Allow To Tag Change Default = Off.
When set to On, allows the IP Office to change media parameters when connecting
a call to a different party than that which was advertised in the media parameters of
provisional responses, such as 183 Session Progress.
P-Early-Media Default = None.
Support
The options are:
• None: IP Office will not advertise support of this SIP header and will always take
incoming early media into account regardless of presence of this header
• Receive: IP Office will advertise support of this SIP header and will discard
incoming early media unless this header is present in the SIP message.
• All: IP Office will advertise support of this SIP header, will discard incoming early
media unless this header is present in the SIP message and will include this SIP
header when providing early media.
Send Default = Off.
SilenceSupp=off
Used for the G711 codec. When checked, the silence suppression off attribute is sent
in SDP on this trunk.
Force Early Direct Default = Off.
Media
When set to On, allows the direct connection of early media streams to IP endpoints
rather than anchoring it at the IP Office.
Media Connection Default = Disabled.
Preservation
When enabled, allows established calls to continue despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
Preservation on public SIP trunks is not supported until tested with a specific service
provider.
Indicate HOLD Default = Off.
When enabled, the system sends a HOLD INVITE to the SIP trunk endpoint.
Media Security Default = Off
When enabled, the IP Office advertises support of this SIP header, to indicate that
audio is configured to be secure and is enforced to use SRTP only. This supports the
SIP security header defined by RFC3329.
This option is available only when:
• TLS is being used.
• Line | SIP Line | VoIP > Media Security is selected and set to Enforced.
• Line | SIP Line | VoIP > Fax Transport Support is not set to T38 or T38 Fallback.
When the configuration file is sent to the system, the SIP trunk is restarted and all
calls on the line are dropped.

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Call Control
Field Description
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
(s)
Sets how long the IP Office system should wait for a response to an attempt to initiate
a call before following the alternate routes set in an ARS form.
Call Queuing Timeout Default = 5 minutes.
(m)
• For incoming calls, this sets how many minutes the IP Office waits before dropping a
call that is waiting for VCM resources or has remained in the unanswered state.
• For outgoing calls, this sets how many minutes the IP Office waits for a call to be
answered after receiving a provisional response.
Service Busy Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).
Response
For calls that result in a busy response from IP Office, this setting determines the
response code. The options are:
• 486 - Busy Here
• 503 - Service Unavailable
on No User Default = 408-Request Timeout.
Responding Send
Specifies the cause to be used when releasing incoming calls from SIP trunks, when
the cause of releasing is that user did not respond. The options are 408-Request
Timeout or 480 Temporarily Unavailable.
Action on CAC Default = Allow Voicemail
Location Limit
When set to Allow Voicemail, the call is allowed to go to a user's voicemail when
the user's location call limit has been reached. When set to Reject Call, the call is
rejected with the failure response code configured in the Service Busy Response
field.
Suppress Q.850 Default = Off.
Reason Header
When SIP calls are released by sending BYE and CANCEL, a release reason header
is added to the message. When set to On, the Q.850 reason header is not included.
Emulate NOTIFY for Default = Off.
REFER
Use for SIP providers that do not send NOTIFY messages. When set to On, after IP
Office issues a REFER, and the provider responds with 202 ACCEPTED, IP Office will
assume the transfer is complete and issue a BYE.
No REFER if using Default = Off.
Diversion
When enabled, REFER is not sent on the trunk if the forwarding was done with 'Send
Caller ID = Diversion Header'. Applies to Forwards and Twinning.

Related links
SIP Line on page 372

Engineering
Navigation: Line | SIP Line | Engineering

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 SIP Line

You can use this tab to enter commands that apply special features to the SIP line. The
commands are called SIP Line Custom (SLIC) strings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
reINVITE Codec Renegotiation
For R11.0 and higher, the IP Office supports codec renegotiation when a reINVITE is received.
See Codec selection on page 857.
You can use the following command to retain the pre-R11.0 behavior of no renegotiation. Note: On
existing IP Office systems upgraded to R11.0 or higher, this command is automatically added to all
existing SIP lines.
• SLIC_PREFER_EXISTING_CODEC
Calling Number Validation
You can use the following commands to control calling number validation. See SIP Calling Number
Verification (STIR/SHAKEN) on page 866.
• SLIC_STIR_REJECT_CODE=<n> where <n> is the response code sent for calls rejected by
the IP Office.
• SLIC_STIR_REJECT_STRING=<y> where <y> is the response string sent for calls rejected
by the IP Office.
• SLIC_STIR_ATTEST="<w>" where <w> is the name of the header the IP Office checks for
a call's authorization level.
• SLIC_STIR_CUSTOM=<z> where <z> value enables or disables various call features.
Server Name Identification (SNI)
The following SLIC codes can be used for SIP trunks using TLS. When used:
• On outgoing connections, the IP Office adds Server Name Indication (SNI) information to the
SAN field it sends.
• If the IP Office system's Received certificate checks (Telephony endpoints) settings is set
to Medium + Remote Checks or High + Remote Checks, then the SLIC value is also used
to validate the received certificates SAN.
The SLIC codes are:
• SLI_ADD_SIP_SAN=<X>
Use a SNI set to sip:<SNI> where the <SNI> value is taken from the existing IP Office SIP
line configuration based on the following values of <X> as below:
- D = Use the value of the SIP line's ITSP Domain Name setting (Line > SIP Line). For
example, for a SIP line with the ITSP Domain Name set to ipo.example.com, adding
SLIC_ADD_SIP_SAN=D sets the SNI added to sip:ipo.example.com.
- P = Use the value of the SIP line's configured ITSP Proxy Address setting (Line >
Transport > ). This option is only supported for a ITSP Proxy Address set to a single
address. For example: SLI_ADD_SIP_SAN=P
Keepalives
Supported with IP Office R11.1.3.1 and higher.

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Line

You can add SLIC_HNT_EMPTY_PACKET to have the SIP line send RTP packets with payload 20
(unassigned payload) and no data as keepalives. This overrides the default of send STUN packets
for keepalives.
Related links
SIP Line on page 372

SM Line
This type of line is used to create a SIP connection between an IP Office and an Avaya Aura®
Session Manager. The other end of the SIP connection must be configured on the Session
Manager as a SIP Entity Link.
An SM Line can only be added to IP Office system Standard Mode or Server Edition
configurations. It is typically used in IP Office Standard mode in Enterprise Branch deployments
connected to the Avaya Aura® network. For more details about IP Office Enterprise Branch
deployments refer to Deploying Avaya IP Office™ Platform as an Enterprise Branch with Avaya
Aura® Session Manager.
An SM Line can also be used in IP Office Server Edition to connect to an Avaya Aura® Session
Manager. Through the SM Line, IP Office Server Edition supports interoperability with Avaya
Aura® Session Manager. It also supports interoperability, via the Avaya Aura® Session Manager,
with Avaya Aura® Communication Manager systems and with CS 1000 systems. Note that IP
Office Server Edition is not used as an enterprise branch product and does not support some
of the IP Office enterprise branch functionality, such as management by Avaya Aura® System
Manager, WebLM licensing, Centralized Users or voicemail over the SM Line.
If the Avaya Aura® network has multiple Avaya Aura® Session Managers to provide redundancy,
two SM lines can be added, one configured for each Avaya Aura® Session Manager.
Related links
Line on page 290
Session Manager on page 400
VoIP on page 403
T38 Fax on page 407

Session Manager
Navigation: Line | SM Line | Session Manager
Additional configuration information
For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 621.
Configuration settings
These settings are not mergeable. Changes to these settings require a reboot of the system.

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 SM Line

Changing the In Service setting to Disabled (out of service) requires a system reboot. However,
changing the In Service setting to Enabled is mergeable. Configuration changes made while the
line is out of service are also mergeable.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
• Session Manager line prioritization: Up to two Session Manager lines can
be configured. The two Session Manager lines are prioritized based on the line
number. The lower line number is considered the primary Session Manager line.
For example, if the first Session Manager line is configured as line number 17
and the second Session Manager line is configured as line 18, then line number
17 is considered the primary Session Manager line. If you want to designate the
second Session Manager line (line 18 in this example) as the primary Session
Manager line, you must change one or both of the line numbers so that the
second Session Manager line is configured with a lower number than the current
primary line.
• Session Manager line redundancy: Based on the priority of the Session
Manager lines designated by the line number, the active line to which the IP
Office> sends all calls will always be the highest priority Session Manager line in
service. That is, if the primary Session Manager line is in service, it will be the
active line for sending calls. If the connection to the primary Session Manager
line is lost, causing the IP Office to switch to the secondary Session Manager
line, then when the primary line comes back up later, the IP Office reverts back
to the primary Session Manager line.
In Service Default = Enabled
This option can be used to administratively disable the SM Line. It does not reflect
the dynamic state of the line. If an SM Line is administratively disabled it is not
equivalent to being in the dynamic out of service state.
SM Domain Name This should match a SIP domain defined in the Session Manager system's SIP
Domains table. Unless there are reasons to do otherwise, all the Enterprise Branch
systems in the Avaya Aura® network can share the same domain.
SM Address Enter the IP address of the Session Manager the line should use in the Avaya
Aura network. The same Session Manager should be used for the matching Entity
Link record in the Avaya Aura® configuration.
Outgoing Group ID Default = 98888
This value is not changeable. However note the value as it is used in Enterprise
Branch short codes used to route calls to the Session Manager.
Prefix Default = Blank
This prefix will be added to any source number received with incoming calls.
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Line

Field Description
Max Calls Default = 10
Sets the number of simultaneous calls allowed between the Enterprise Branch and
Session Manager using this connection. Each call will use one of the available
licenses that are shared by all SIP trunks configured in the system.
Network Type Default = Public.
This option is available when System | Telephony | Telephony | Restrict
Network Interconnect is enabled. It allows you to configure trunks as either
Public or Private.
• The IP Office will return number busy indication to any attempt to connect a call
on a Private trunk to a Public trunk or the opposite.
• The call restriction includes transfers, forwarding and conference calls.
• Avaya does not recommended use of this feature on IP Office systems using
any of the following features: multi-site networks, VPNremote, application
telecommuter mode.
Include location specific Default = Off.
information
Enabled when Network Type is set to Private. Set to On if the PBX on the other
end of the trunk is toll compliant.
URI Type Default = SIP.
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, [email protected]). This affects the From field of outgoing calls.
The To field for outgoing calls will always use the format specified by the short
codes used for outgoing call routing. Recommendation: When SIP Secured URI is
required, the URI Type should be set to SIPS. SIPS can be used only when Layer
4 Protocol is set to TLS.
Media Connection Default = Enabled.
Preservation
When enabled, attempts to maintain established calls despite brief network
failures. Call handling features are no longer available when a call is in a
preserved state. When enabled, Media Connection Preservation applies to Avaya
H.323 phones that support connection preservation.
Location
Network Configuration
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Layer 4 Protocol Default = TCP.
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 SM Line

Field Description
Send Port When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.
Listen Port When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.

Session Timer Default = 1200. Range = 90 to 64800

This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. Setting the field to On Demand disables the
session timer.
Communication Manager supports SIP session refresh via UPDATE in
Communicaton Manger release 6.2 SP1 and later. If using an earlier release of
Communication Manager, then the Session Timer parameter must be set to On
Demand.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The
description is not used elsewhere.

Related links
SM Line on page 400

VoIP
Navigation: Line | SM Line | VoIP
These settings are mergeable. Changes to these settings do not require a reboot of the system.
These settings can be edited online. Changes to these settings do not require a reboot of the
system.

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Line

Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend
all systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
Fax Transport Support Default = None.
This option is available only if Re-Invite Supported is selected.
• IP500 V2 systems can terminate T38 fax calls.
• Linux-based IP Office systems can route the calls between trunks/terminals with
compatible fax types.
• Set the method the IP Office uses to handle fax calls.
The supported options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does
not support T38, the IP Office will send a re-invite to change the transport
method to G.711.
Call Initiation Timeout (s) Default = 4 seconds. Range = 1 to 99 seconds.
Sets how long the IP Office system should wait for a response to an attempt to
initiate a call before following the alternate routes set in an ARS form.
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 SM Line

Field Description
DTMF Support Default = RFC2833 (IP500 V2), RFC2833/RFC4733 (Linux-Based Server)
Selects the method the IP Office uses to signal DTMF key press digits to the
remote end. The options are:
• In Band - Send digits as part of the audio path.
• RFC2833 or RFC2833/RF4733 - Send digits using a separate audio stream from
the voice path. If not supported by the far end, the line reverts to using In Band
signaling.
• Info - Send the digits in SIP INFO packets.
Media Security Default = Same as System.
These setting control whether SRTP is used for this line and the settings used for
the SRTP. The options are:
• Same as System: Matches the system setting at System | System | VoIP |
VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) is enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and
data) is enforced to use SRTP only. Selecting Enforced on a line or extension
that does not support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on
System | System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be
protected using encryption. The default is to encrypt just the RTP stream (the
speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be
protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
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Line

Field Description
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any
audio data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
Allow Direct Media Path Default = On
This settings controls whether calls between IP endpoints and/or lines must go
through the IP Office or can try to route directly if possible within the customer
network.
• If disabled, calls go through the IP Office and use its resources. RTP relay
support may allow calls between devices using the same audio codec to not
require a voice compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both
ends must support direct media and have matching VoIP settings, for example
using the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so
on. Otherwise, the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt
direct media even if the other end has differing DTMF settings.
Re-Invite Supported Default = Off.
When enabled, the IP Office can use Re-Invite during a call to change the
characteristics of the call. For example, when the target of an incoming call or a
transfer does not support the codec originally negotiated on the trunk.
• Requires the ITSP to also support Re-Invite.
• This setting must be enabled for video support.
Codec Lockdown Default = Off.
In response to a SIP offer with a list of codecs, some SIP user agents send a SDP
answer that also lists multiple codecs. The user agent can then switch to any of
those codecs during the session without requiring further negotiation. However, IP
Office does not support this, so loss of speech path occurs if the current codec
changes without renegotiation.
• If enabled, when the IP Office receives an SDP answer with multiple codecs from
its list of offered codecs, the IP Office sends a re-INVITE using just a single
codec from the list, and an SIP offer with just the single chosen codec.
• This option requires Re-Invite Supported enabled.
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 SM Line

Field Description
Force direct media with Default = On
phones
When enabled, if an Avaya IP phone dials digits during a direct media call, the
IP Office changes the call to indirect media and sends the digits as RFC2833.
15-seconds after the last digit, the IP Office changes the call back to direct media.
• This setting is requires the line to have Re-Invite Supported and Allow Direct
Media Path enabled, and DTMF Support set to RFC2833/RF4733.
G.711 Fax ECAN Default = Off
When enabled, if the IP Office detects a fax call, it switches to G.711 with echo
cancellation (ECAN) based on the 'G.711 Fax ECAN field, NLP disabled, a fixed
jitter buffer, and silence suppression is disabled. You can use this to avoid an
ECAN mismatch with the trunk provider.
• This setting is only available on IP500 V2 systems when Fax Transport Support
is set to G.711 or T38 Fallback.

Related links
SM Line on page 400

T38 Fax
Navigation: Line | SM Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which they
both support. The options are: 0, 1, 2, 3.
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For UDPTL,
redundancy error correction is supported. Forward Error Correction (FEC) is not
supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed V.21
T.30 fax transmissions.
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Line

Field Description
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27 and
V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax equipment or is
found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Default = 150.
Timeout (secs)
Scan Line Fix-up Default = On.
TFOP Default = On.
Enhancement
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression
NSF Override Default = Off.
If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be
overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
SM Line on page 400

S0 Line

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 S0 Line

These settings are used for S0 ports provided by an S08 expansion module connected a control
unit. For full details of installation refer to the IP Office Installation manual.
Though displayed as lines, these BRI ports are used for connection of ISDN2 devices such as
video conferencing units or ISDN PC cards.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
Related links
Line on page 290
S0 Line on page 409
S0 Short Codes on page 411
Line | S0 Channels on page 411

S0 Line
Navigation: Line | S0 Line
The following settings are not mergeable. Changes to these settings require a system reboot.
• Line Sub Type
• Network Type
The remaining settings are mergeable.
Field Description
Line Number This parameter is not configurable. It is allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing the
line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Line Sub Type Default = ETSI Select to match the particular line type provided by the line provider.
Network Type Default = Public.
This option is available if Restrict Network Interconnect (System | Telephony |
Telephony) is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a Private
trunk to a Public trunk or vice versa. This restriction includes transfers, forwarding and
conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote, application
telecommuter mode.
Telephone Number Used to remember the telephone number of this line. For information only.
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Line

Field Description
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a number
is presented from ISDN as an "international number" this prefix is added. For example
441923000000 is converted to 00441923000000.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
When a short code specifies a number to dial, the IP Office will seize an available line
from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network.
Reserved Group ID Numbers:
• 0 - In a Server Edition/Select network, the ID 0 cannot be used.
• 90000 - 99999 - Reserved for system use (not enforced).
- 96666 - Use for ACO lines.
- 98888 - For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
- 99001 - 99148 - In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
- 99998 - In a Server Edition/Select network, reserved for the IP Office lines to the
secondary server.
- 99999 - In a Server Edition/Select network, reserved for the IP Office lines to the
primary server.
TEI Default = 0
Not used. The Control Unit will ignore any entry.
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 S0 Line

Field Description
Number of Default = 2
Channels
Defines the number of operational channels that are available on this line. 2 for BRI and
up to 30 for PRI - depending upon the number of channels subscribed.
Outgoing Channels Default = 2
This defines the number of channels available, on this line, for outgoing calls. This should
normally be the same as Number of Channels field, but can be reduced to ensure
incoming calls cannot be blocked by outgoing calls.
Voice Channels Default = 2
The number of channels available for voice use.
Data Channels Default = 2
The number of channels available for data use. If left blank the value is 0.

Related links
S0 Line on page 408

S0 Short Codes
Navigation: Line | S0 Line | Short Codes
For BRI S0 lines , these settings are mergeable.
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those
types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Related links
S0 Line on page 408

Line | S0 Channels
Navigation: Line | S0 Line | Channels
For S0 channels this form is not used.

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Line

Related links
S0 Line on page 408

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Chapter 27: Control Unit

Navigation: Control Unit | Unit

The Control Unit menu displays details of the IP Office system.
• For Linux-based IP Office systems, the group view lists menu shows details for the physical
server platform and the IP Office services hosted on the server.
• For the IP500 V2 control units, the menu shows details of the control unit software version and
the cards installed into the control unit.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
Device Number This number is automatically allocated by the IP Office system.
Unit Type The name of the device.
Version The version of software.
Serial Number For internal components, this is the MAC address of the IP Office server or control
unit.
Unit IP Address For the IP Office control unit, this field shows the IP address for LAN1.
Interconnect Number For IP500 V2 external expansion modules, this is the control unit expansion port
used for connection. For other devices, this is 0.
Module Number For IP500 V2 external expansion modules, this is the control unit expansion port
used for connection. Control Unit is displayed for internal devices and services.
Operating Mode This field is shown for DS16B or DS30B digital expansion modules. Select the
operating mode based on the type of telephones deployed.
• DS - 1400, 9500 Series Phones
• BST - T7000, M7000 Series Phones

The New and Delete actions on this form have special functions.
Field Description
New You can use this option to add a WAN3 external expansion module if the module is
not detected following a system reboot.
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Control Unit

Field Description
Delete You can use this action to remove details of an external expansion module from the
configuration.
• Deleting an extension or line module also deletes any extensions or lines
associated with the module.
• If the external expansion module is still physically attached, following a reboot the
IP Office automatically creates new default extension or line entries.

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Chapter 28: Extension

By default, each extension is normally associated with a user and uses that user's directory number
and other settings. Users with a log in code can move between extensions by logging in and out, so
the directory number is not a fixed property of the extension.
Non-IP Extensions
Physical extension ports are either integral to the control unit or added by the installation of an
analog or digital phone expansion module. Extension records are automatically created for each
physical extension port within the system. These ports cannot be added or deleted manually. For
Server Edition, non-IP extensions are only supported on Expansion System (V2) units.
Icon Description
Standard Telephone - A standard extension.

Quiet Headset - Used for analog extension devices that are permanently off-hook.

IVR Port - Used for analog ports connected to devices that require a specific disconnect clear signal
at the end of each call.
Paging Speaker - An analog extension port set to be used as a paging speaker connection.

FAX Machine - Indicates that the extension is connected to a FAX machine.

MOH Source - Indicates that the extension is being used as a music on hold source.

IP Extensions
These are used for IP phone devices and VoIP applications.
Icon Description
H.323 or SIP Extension - This icon indicates an IP extension. IP extensions are either added
manually or by the automatic detection of the phone being connected. IP extensions can also be
added manually to support a third-party IP phone device. Note that third-party IP phone devices
require entry of an IP End-Points license.
IP DECT or SIP DECT - An extension port manually added to match extensions within an Avaya IP
DECT system connected to the system via an IP DECT line.

Related links
Extn on page 416
Analog on page 419

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Extension

Extension VoIP on page 421

Extension T38 Fax on page 429
IP DECT Extension on page 431
SIP DECT Base Extension on page 432

Extn
Navigation: Extension | Extn
Additional configuration information
The Caller Display Type setting controls the presentation of caller display information. For
additional configuration information, see Caller Display on page 617.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Configuration Settings
These settings are mergeable except Base Extension and Caller Display Type which require a
system reboot.
Field Description
Extension ID The physical ID of the extension port. Except for IP extensions, this settings is allocated
by the system and is not configurable.
Base Extension Range = 2 to 15 digits.
This is the directory number of the extension's default associated user if one is required.
• The field can be left blank for digital and analog extensions, creating and extension
where users are forced to login but the extension has no default associated user. This
option is not supported for IP and CTI extensions.
• Following a restart, the system attempts to log in the user with the same extension
number if they are not already logged in elsewhere in the multi-site network. This does
not occur if that user is set to Force Login.
• If another user logs onto an extension, when they log out, the extension returns to
its default associated user unless they have logged in elsewhere or are set to Force
Login.
Phone Password Default = Blank. Range = 9 to 13 digits.
H.323 and SIP extensions only. This password is entered as part of phone registration
with the IP Office system.
Caller Display Type Default = On.
Controls the presentation of caller display information for analog extensions. For digital
and IP extensions, this value is fixed as On. The table below lists the supported options,
all others are currently not used and default to matching UK.
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 Extn

Field Description
Type Description
Off Disables caller display.
On Enables caller display using the caller display type appropriate to the System
Locale, see Avaya IP Office Locale Settings. If a different setting is required
it can be selected from the list of supported options. For an analog extension
connected to a fax server or other device that requires the pass through of
DTMF tones, select DTMFF.
UK FSK before the first ring conforming to BT SIN 227. Name and number.
UK20 As per UK but with a maximum length of 20 characters. Name and number.
DTMFA Caller ID in the DTMF pattern A<caller ID>C. Number only.
DTMFB Caller ID in DTMF after call connection. Number only.
DTMFC Caller ID in the DTMF pattern A<caller ID>#. Number only.
DTMFF Sends the called number in DTMF after call connection. Number only. Used
for fax servers. When calls are delivered via a hunt group it is recommended
that hunt group queuing is not used. If hunt group queuing is being used, set
the Queue Type to Assign Call on Agent Alert.
DTMFD Caller ID in the DTMF pattern D<caller ID>C. Number only.
FSKA Variant of UK used for BT Relate 1100 phones. Name and number.
FSKB ETSI specification with 0.25 second leading ring. Name and number.
FSKC ETSI specification with 1.2 second leading ring. Name and number.
FSKD Conforms to Belcore specification. Name and number.
Reset Volume after Default = Off.
Calls
Resets the phone's handset volume after each call. This option is supported on Avaya
1400, 1600, 2400, 4400, 4600, 5400, 5600, 6400, 9500 and 9600 Series phones.
Device Type This field indicates, the last known type of phone connected to the extension port.
• Analog extension ports always report as Analog Handset since the presence or
absence of actual analog phone cannot be detected.
• Digital extension ports report the type of digital phone connected or Unknown digital
handset if no phone is detected.
• H.323 extensions report the type of IP phone registered or Unknown H.323 handset if
no phone is currently registered as that extension.
• SIP extensions report the type of SIP phone registered or Unknown SIP device if
no SIP device is currently registered as that extension. Applications such as Avaya
Workplace Client and one-X Mobile Preferred that do not use extension records also
display Device type as Unknown SIP device.
For some types of phone, the phone can only report its general type to the system but
not the specific model. When that is the case, the field acts as a drop-drown to select
a specific model. The value selected is also reported in other applications such as the
System Status Application, SNMP, etc.
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Extension

Field Description
Default Possible Phone Models
Type
T7100 M7100, M7100N, T7100, Audio Conferencing Unit.
T7208 M7208, M7208N, T7208.
M7310 M7310, M7310N, T7406, T7406E.
M7310B M7310BLF, T7316.
LF
M7324 M7324, M7324N.
Location The drop down list contains all locations that have been defined on the system: Location
| Location. See Using Locations on page 617.
Associating an extension with a location:
• Allows emergency call routing using settings specific to that location.
• Allows the display of location based time. Supported on 1100, 1200, 1600, 9600 and
J100 Series phones and D100, E129 and B179 telephones.
• For DECT R4 extensions, the extension location can be overridden on a call-by-call
basis using the location name specified in the base station configuration. Supported
with R11.1 FP2 SP2 and higher. Requires Call based Location Information to be set
on the IP DECT line and each base station to be configured with a location name that
matches one in the IP Office configuration.
Fallback as Default = Auto.
Remote Worker
Determines what fallback address is used for Remote Worker phone resiliency.
The options are:
• Auto: Use the fallback address configured on the IP Office Line providing the service.
• No: Use the alternate gateway private address.
• Yes: Use the alternate gateway public address.
Module This field indicates the external expansion module on which the port is located. BP
indicates an analog phone extension port on the base or control unit. BD indicates a
digital station (DS) port on the control unit. For an IP500 V2 control unit, BD and BP is
also followed by the slot number. VoIP extensions report as 0.
Port This field indicates the port number on the Module indicated above. VoIP extensions
report as 0.
Disable Default = Off (Speakerphone enabled).
Speakerphone
When selected, disables the fixed SPEAKER button if present on the phone using this
extension port. Only supported on Avaya DS, TCM and H.323 IP phones. An audible
beep is sounded when a disabled SPEAKER button is pressed. Incoming calls such
as pages and intercom calls are still connected but the speech path is not audible until
the user goes off-hook using the handset or headset. Similarly calls made or answered
using other buttons on the phone are not audible unless the user goes off-hook using the
handset or headset. Currently connected calls are not affected by changes to this setting.

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 Analog

Related links
Extension on page 415

Analog
Navigation: Extension | Analog Extension | Analog
This tab contains settings that are applicable to analog extensions. These extensions are provided
by ports marked as POT or PHONE on control units and expansion modules.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Equipment Classification:
Field Description
Default = Standard Telephone.
Only available for analog extension ports. Note that changes to this setting are mergeable.
Quiet Headset On extensions set to Quiet Headset, the audio path is disabled when the extension is
idle. Ringing is presented in the audio path. Caller ID is not supported on the phone.
This option can be used with analog extensions where the handset is replaced by a
headset and all audio, including ringing should be through the headset.
Since the audio path is disabled when idle, the Quiet Headset extension cannot dial
digits to make calls. Therefore, to make and answer calls this option is typically used with
the user Offhook Station (User > Telephony > Call Settings setting which allows the
extension user to make and answer calls using applications.
Paging Speaker Used for analog ports connected to a paging amplifier. This extension will present busy
and cannot be called or be used to make calls. It can only be accessed using Dial Paging
features.
When using a UPAM connected to an analog extension port, the extension's Equipment
Classification should be set to IVR Port rather than Paging Speaker.
Standard Use for normal analog phones.
Telephone
Door Phone 1/Door These two options are currently not used and so are grayed out.
Phone 2
IVR Port Used for analog ports connected to devices that require a disconnect clear signal (a
break in the loop current) at the end of each call. When selected the Disconnect Pulse
Width is used.
FAX Machine If fax Relay is being used, this setting should be selected on any analog extension
connected to an analog fax machine. This setting can also be used with SIP trunks.
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Extension

Field Description
MOH Source If selected, the port can be used as a music on hold source in the System >
Telephony > Tones and Music settings. An extension set as a music on hold source
cannot make or receive calls. The audio input can be monitored through the extension
music on hold controls.
A suitable interface device is required to provide the audio input to the extension port.
It must look to the system like an off-hook analog phone. For example, a transformer
with a 600 Ohm winding (such as a Bogen WMT1A) or a dedicated MoH device with a
600 Ohm output designed for connection to a PBX extension port which is providing loop
current can be used.

Flash Hook Pulse Width

The following options are only available for analog extension ports. They define the length of loop
break that will be considered a time break recall (TBR) signal.
Field Description
Use System Default = On
Defaults
Use the default values appropriate to the system's locale. Refer to Avaya IP Office
Locale Settings.
Minimum Width Range = 20 to 2540 milliseconds.
Minimum hook flash length used if Use System Defaults is not selected. Shorter breaks
are ignored a glitches.
Maximum Width Range = 30 to 2550 milliseconds.
Maximum hook flash length used if Use System Defaults is not selected. Longer breaks
are treated as clearing.
Disconnect Pulse Default = 0ms. Range = 0 to 2550ms
Width
This setting is used with analog extensions where the Equipment Classification above
has been set to IVR Port. It sets the length of loop current break used to indicate call
clearing.

Message Waiting Lamp Indication Type

Field Description
Message Waiting Default = None
Lamp Indication
Allows the selection of the message waiting indication (MWI) mode for analog and IP
Type
DECT extensions. The options are: On (See below), 51V Stepped, 81V, 101V (Phone
V2 and IP500 Phone base cards), Bellcore FSK, Line Reversal A, Line Reversal B.

If the option Restrict Analog Extension Ringer Voltage is selected (System | Telephony |
Telephony), the MWI options are restricted to: Line Reversal A, Line Reversal B or None with
the default Line Reversal A.
On defaults the message waiting indication setting as follows based on the system locale:

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 Extension VoIP

Setting Locale
51V Stepped Argentina, Australia, Brazil, Canada, Chile, China, Colombia, Japan, Korea, Mexico, New
Zealand, Peru, Russia, Saudi Arabia, South Africa, Spain, United States, Venezuela
101V on Phone Bahrain, Belgium, Denmark, Egypt, Finland, France, Germany, Greece, Hong Kong,
V2 modules and Hungary, Iceland, Italy, India, Kuwait, Morocco, Netherlands, Norway, Oman, Pakistan,
IP500 Phone cards, Poland, Portugal, Qatar, Singapore, Sweden, Switzerland, Taiwan, Turkey, United Arab
otherwise 81V. Emirates, United Kingdom

Hook Persistency
Field Description
Hook Persistency Default = 100ms. Range = 50 to 255ms.
Sets the minimum time the extension needs to be off-hook before the system treats it as
off-hook and applies any off-hook features. For example dialing timers or hot-dial short
codes.
Shorter periods of off-hook time are ignored.

Related links
Extension on page 415

Extension VoIP
This tab is only available for H.323 and SIP extensions. The settings available will vary depending
on the extension type.
Related links
Extension on page 415
Extension H.323 VoIP on page 421
SIP Extension VoIP on page 425

Extension H.323 VoIP

Navigation: Extension | H323 Extension | VoIP
These settings are shown for a H.323 IP extension.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
IP Address Default = 0.0.0.0 (Accept any address)
If an address is entered here, the IP Office will only accept registration from a device
with that address.
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Extension

Field Description
MAC Address Default = 0000000000000 (Grayed out)
This field is grayed out and not used.
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
Supplementary Default = H450.
Services
Selects the supplementary service signaling method for use with non-Avaya IP devices.
Options are None, QSIG and H450. For H450, hold and transfer are supported. Note
that the selected method must be supported by the remote end.
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 Extension VoIP

Field Description
Media Security Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Same as System: Matches the system setting at System | System | VoIP | VoIP
Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data)
is enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Advanced Media Default = Same as System.
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same settings as the system setting configured on
System | System | VoIP | VoIP Security.
• Encryptions: Default = RTP
This setting allows selection of which parts of a media session should be protected
using encryption. The default is to encrypt just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP
This setting allows selection of which parts of the media session should be protected
using authentication.
• Replay Protection SRTP Window Size: Default = 64. Not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.
There is also the option to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any audio
data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
Enable FastStart Default = Off
for non-Avaya IP
A fast connection procedure. Reduces the number of messages that need to be
Phones
exchanged before an audio channel is created.
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Extension

Field Description
Out of Band DTMF Default = On
When on, DTMF is sent as a separate signal ("Out of Band") rather than as part of the
encoded voice stream ("In Band"). The "Out of Band" signaling is inserted back into
the audio by the remote end. This is recommended for low bit-rate compression modes
such as G.729 and G.723 where DTMF in the voice stream can become distorted.
For Avaya 1600, 4600, 5600 and 9600 Series phones, the system will enforce the
appropriate setting for the phone type.
Requires DTMF Default = Off.
This field is displayed when Ignore DTMF Mismatch for Phones (System > > VoIP is
enabled.
• If disabled, during the checks for direct media on calls between two VoIP phones, the
system ignores the DTMF checks.
- Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, do not match.
• If enabled, the extension needs to receive DTMF signals.
- When enabled, the extension setting is ignored for contact center applications.
Contact center application SIP extensions are always treated as requiring DTMF.
Local Tones Default = Off
When selected, the H.323 phones generate their own tones.
Allow Direct Media Default = On
Path
This settings controls whether calls between IP endpoints and/or lines must go through
the IP Office or can try to route directly if possible within the customer network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a voice
compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both ends
must support direct media and have matching VoIP settings, for example using the
same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise,
the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.
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 Extension VoIP

Field Description
Reserve License Default = None.
Avaya IP phones require an Avaya IP Endpoint license, non-Avaya IP phones require
an 3rd Party IP Endpoint license. Normally the IP Office issues licenses in the order
that devices register. This option allows this extension to be pre-licensed before the
device registers. This can prevent a previously licensed phone becoming unlicensed
following a system restart. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note:
• When WebLM licensing is enabled, this field is automatically set to Reserve Avaya
IP Endpoint License. The Both and None options are not available.

Related links
Extension VoIP on page 421

SIP Extension VoIP

Navigation: Extension | SIP Extension | VoIP
These settings are shown for SIP IP extensions. For example for J100 Series phones.
Field Description
IP Address Default = 0.0.0.0
The IP address of the phone. If an address is entered, the IP Office only accept
registration from a device with that address.
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Extension

Field Description
Reserve License Default = None.
Avaya IP phones require an Avaya IP Endpoint license, non-Avaya IP phones require
an 3rd Party IP Endpoint license. Normally the IP Office issues licenses in the order
that devices register. This option allows this extension to be pre-licensed before the
device registers. This can prevent a previously licensed phone becoming unlicensed
following a system restart. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note:
• When WebLM licensing is enabled, this field is automatically set to Reserve Avaya
IP Endpoint License. The Both and None options are not available.
• When the Profile of the corresponding user is set to Centralized User, this field is
automatically set to Centralized Endpoint License and cannot be changed.
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any audio
data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
Fax Transport: Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
detected, renegotiates the call codec as configured below. The SIP line provider must
support the selected fax method and Re-Invite.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None - Select this option if fax is not supported by the line provider.
• G.711 - Use G.711 to send and receive faxes.
• T38 - Use T38 to send and receive faxes.
• T38 Fallback - Use T38 to send and receive faxes. If the call destination does not
support T38, the IP Office will send a re-invite to change the transport method to
G.711.
DTMF Transport Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The supported options are In Band, RFC2833 or Info.
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 Extension VoIP

Field Description
Requires DTMF Default = Off.
This field is displayed when System Settings > System > VoIP > Ignore DTMF
Mismatch for Phones is set to On. It can be used to allow direct media connections
between devices despite the devices having differing DTMF setting.
When Requires DTMF is set to Off, during the checks for direct media, the system
ignores the DTMF checks if the call is between two VoIP phones. Set to On if the
extension needs to receive DTMF signals.
SIP endpoints using simultaneous login, which do not have physical extensions in the
configuration, are treated by the system as not requiring DTMF.

Note:
• Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
Local Hold Music Default = Off.
When enabled, the extension plays local music when on HOLD.
If System Settings > Line > Add/Edit Trunk Line > SIP Line > SIP Advanced >
Local Hold Music is enabled, the extension Local Hold Music must be disabled to
play far end music to the extension.
Allow Direct Media Default = On
Path
This settings controls whether calls between IP endpoints and/or lines must go through
the IP Office or can try to route directly if possible within the customer network.
• If disabled, calls go through the IP Office and use its resources. RTP relay support
may allow calls between devices using the same audio codec to not require a voice
compression channel.
• If enabled, calls can take routes other than through the IP Office system. Both ends
must support direct media and have matching VoIP settings, for example using the
same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise,
the call goes through the IP Office system.
- For extensions, disabling Requires DTMF allows the extension to attempt direct
media even if the other end has differing DTMF settings.
VoIP Silence Default = Off
Suppression
When selected, if the IP Office detects silence during a call, it does not send any audio
data.
• This feature is not used on IP lines using G.711 between IP Office systems.
• On trunks between networked IP Office systems, you must enabled the setting at
both ends.
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Extension

Field Description
Codec Lockdown Default = Off.
In response to a SIP offer with a list of codecs, some SIP user agents send a SDP
answer that also lists multiple codecs. The user agent can then switch to any of those
codecs during the session without requiring further negotiation. However, IP Office
does not support this, so loss of speech path occurs if the current codec changes
without renegotiation.
• If enabled, when the IP Office receives an SDP answer with multiple codecs from its
list of offered codecs, the IP Office sends a re-INVITE using just a single codec
from the list, and an SIP offer with just the single chosen codec.
• This option requires Re-Invite Supported enabled.
3rd Party Auto Default = None.
Answer
This setting applies to 3rd party standard SIP extensions. The options are:
• RFC 5373: Add an RFC 5373 auto answer header to the INVITE.
• answer-after: Add answer-after header.
• device auto answers: IP Office relies on the phone to auto answer calls.
Media Security Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Same as System: Matches the system setting at System | System | VoIP | VoIP
Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data)
is enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) is
enforced to use SRTP only. Selecting Enforced on a line or extension that does not
support media security results in media setup failures
- Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Table continues…

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 Extension T38 Fax

Field Description
Codec Selection Default = System Default
Set the supported codecs. Within a network of IP Office systems, we recommend all
systems and lines use the same codecs. The options are:
• System Default - Use the codec list set in the system settings.
• Custom - Configure a list of codec preferences for the line.
- You can move codecs between the Unused and Selected set, and change the
order of the selected codecs.
- The codecs available are set by System | System | VoIP | VoIP. The possible
codecs are:
• OPUS - Supported on Linux-based IP Office systems only.
• G.711 ALAW/G.711 ULAW
• G.729
• G.723.1 - Supported on IP500 V2 systems only.
• G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2
systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.

Related links
Extension VoIP on page 421

Extension T38 Fax

Navigation: Extension | SIP Extension | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which they
both support. The options are: 0, 1, 2, 3.
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For UDPTL,
redundancy error correction is supported. Forward Error Correction (FEC) is not
supported.
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Extension

Field Description
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed V.21
T.30 fax transmissions.
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27 and
V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax equipment or is
found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Default = 150.
Timeout (secs)
Scan Line Fix-up Default = On.
TFOP Default = On.
Enhancement
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression
NSF Override Default = Off.
If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be
overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
Extension on page 415

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 IP DECT Extension

IP DECT Extension
Navigation: Extension | IP DECT Extension
IP DECT extensions are created manually after an IP DECT line has been added to the
configuration or added automatically as DECT handsets subscribe to the DECT system.
These settings are mergeable with the exception of the Reserve License setting. Changing the
Reserve License settings requires a reboot of the system.
Field Description
DECT Line ID Use the drop-down list to select the IP DECT line from the system to the Avaya IP
DECT system.
Message Waiting Default = On
Lamp Indication
Allows selection of the message waiting indication to use with the IP DECT extension.
Type
The options are:
• None
• On
Reserve License Default = None.
Avaya IP phones require an Avaya IP Endpoint license in order to register with the
system. Normally licenses are issued in the order that devices register. This option
allows this extension to be pre-licensed before the device has registered. The options
are
• Reserve Avaya IP Endpoint License
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License and cannot be changed.

The additional fields below depend on whether the IP DECT line has Enable Provisioning
selected.
Enable Provisioning Not Selected
Field Description
Handset Type Default = Unknown
Correct selection of the handset type allows application of appropriate settings for
the handset display and buttons. Selectable handset types are supported 3700 Series
phones or Unknown.

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Extension

Enable Provisioning Selected

Field Description
IPEI Default = 0 (Any IPEI)
If set to a value other than 0, sets the IPEI number of the handset that is able to
subscribe to the DECT R4 system using this extension number. The IPEI for each
DECT handset is unique.
Use Handset Default = Off.
Configuration
If Use Handset Configuration. is selected, the handset user is able to set the phone
language and date/time format. If not selected, those settings will be driven by the
system or user locale settings in the system configuration.

Related links
Extension on page 415

SIP DECT Base Extension

Navigation: Extension | SIP DECT Base
This tab is displayed for SIP DECT extensions. These settings are not mergeable. Changes to
these settings require a reboot of the system.
Field Description
SIP DECT Line Use the drop-down list to select the SIP DECT Line from the system to the DECT base
station.

Related links
Extension on page 415

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Chapter 29: User

Additional configuration information

This section provides the User field descriptions.
For additional configuration information, see Configure User Settings on page 713.
Related links
User on page 433
Voicemail on page 442
DND on page 448
Short Codes on page 448
Source Numbers on page 449
Telephony on page 450
Forwarding on page 461
Dial In on page 465
Voice Recording on page 465
Button Programming on page 466
Menu Programming on page 466
Mobility on page 468
Group Memberships on page 472
Announcements on page 472
SIP on page 474
Personal Directory on page 474
User Portal on page 476

User
Navigation: User | User
Additional configuration information
• For a summary of user management, including a description of centralized users, see User
Management Overview on page 713.

This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.

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User

Users are the people who use the system or are Dial In users for data access. A user does not
have to have an extension number that physical exists - this is useful if the user do not require a
physical extension but wish to use system features.
• The NoUser user is used to apply settings to extensions which have no associated user. Do
not delete this user/
• The Remote Manager user is used as the default settings for dial in connections.
Configuration Settings
You can merge these settings without needing to reboot the IP Office.
• Except adding/removing centralized branch users which requires a system reboot.
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
Field Description
Name Range = Up to 15 characters.
This is the user's account name used for RAS Dial In, caller display and voicemail
mailbox. As the display on caller display telephones is normally 16 characters, it is useful
to keep the name short.
• Only alphanumeric characters and space are supported in this field.
• Names should not start with a space.
• Do not use punctuation characters such as #, ?, /, ^, > and ,.
• This field is case sensitive and must be unique.
• If the IP Office system includes voicemail:
- Voicemail uses the name to create a matching user mailbox. Changing a user's name
will routes their voicemail calls to a new mailbox.
- Voicemail Pro is not case sensitive. It treats names such as "Steve Smith", "steve
smith" and "STEVE SMITH" as all being the same user.
• If the IP Office system includes Avaya one-X Portal:
- Do not use the Name "admin". That user name is a reserved value for Avaya one-X
Portal use.
- Do not use names that include a _ character.
Authentication Default = Blank. Range = Up to 31 alphanumeric characters.
Name
Used on an IP500 V2 system configured as an Avaya Cloud Office™ gateway. See the
Deploying an IP Office as an Avaya Cloud Office ATA Gateway manual.
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 User

Field Description
Password Default = Blank. Range = Up to 31 alphanumeric characters.
This password is used by user applications such as IP Office SoftConsole, User Portal
and TAPI. It is also used for users with Dial In access.
• Note that this is not the user's voicemail mailbox password (see User | Voicemail
| Voicemail Code) or their phone log in code (see User | Telephony | Supervisor
Settings | Login Code).
• The password complexity rules are set through the IP Office security settings. IP
Office Manager will display an error if the password does not meet the complexity
requirements. However, IP Office Manager will still allow you to save the configuration
(unless the system locale is set to France2).
Unique Identity Default = Blank.
An email address for the user. The address must be unique for each user. This email
address is used for:
• Avaya Spaces/Avaya Workplace Client login.
- When used in these roles, for pre-R11.1.2 systems, the unique identity is limited to 15
characters maximum before the @ character.
• Gmail voicemail to email messages.
This setting is separate, though it can be the same address, from the user’s Email
Address setting (see below) which is used for other email functions such as voicemail
email.
Login Code Default = Blank. Range = Up to 31 digits.
Confirm Login • Login code must be at least 4 digits for DS port users.
code
• Login codes of up to 15 digits are supported with Extn Login buttons.
• Login codes of up to 31 digits are supported with Extn Login short codes.
This code is used for logging in on a phone (and for restricting access to features on
phones. See Hot Desking on page 758.
• Hot desking is not supported for centralized users. Centralized users use the Login
Code for SIP registration on Session Manager.
• Normally users can only log out if they have a Login Code set or if they are currently
logged in at an extension whose Base Extension number no longer matches their own
Extension setting.
• When set, the short code feature Change Login Code can be used by users to change
their login code.
• If the user has a login code set, it is used by the Outgoing Call Bar Off short code
feature.
• If the user has a login code set, access to a range of programmable button features
requires entry of the login code. For example, access Self Admin and System Phone
features.
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User

Field Description
Audio Conference Default = Blank. Range = Up to 15 numeric characters.
PIN
Use this field to configure PIN access for meet me conferences.
• An L in this field disabled the unscheduled meet-me conference feature for the user.
Account Status Default = Enabled.
Use this setting set the user account to Enable, Disable, or Force New Password.
• When set to Force New Password, the user can only set a new password by logging in
using Avaya one-X Portal.
The IP Office system can change if they make too many failed log in attempts. This uses
settings configured in the IP Office security setting:
• If a user exceeds the Password Reject Action, then the Password Reject Action is
implemented.
- If the Password Reject Action is Log and Disable Account, then the account
status is changed to Locked - Password Error.
- If the Password Reject Action is Log and Temporary Disable, then the account
status is changed to Locked - Temporary.
Full Name Default = Blank
Use this field to enter the user's full name. When set, the Full Name is used in place of
the Name for display by phones and user applications.
• Names should not start with a space.
• Do not use punctuation characters such as @, #, ?, /, ^, > and ,.
• The recommended format is <first name><space><last name> for the name to be used
correctly by voicemail dial by name features.
Extension Range = 2 to 15 digits.
In general all extensions should have the same number of digits. This setting can be left
blank for users used just for dial in data connections.
• Users associated with IP phones or who may log in as such devices should not be
given extension numbers greater than 7 digits.
• Centralized users’ extension numbers can be up to 13 digits in length. Although IP
Office supports extension numbers up to 15 digits, the 13-digit length is determined by
the maximum extension number length allowed for provisioning Centralized users in
Communication Manager.
Email Address Default = Blank
This address is used as the user’s email address for a range of functions. Primarily it is
used for voicemail-email functions if required. It is also used for any other emails that the
system may send to the user.
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 User

Field Description
Locale Default = Blank (Use system locale)
Configures the language used for voicemail prompts played to the user, assuming the
language is available on the voicemail server. See Avaya IP Office Locale Settings. On a
digital extension it also controls the display language used for messages from the system.
Note however that some phones have their own menu options for the selected language
for the phone menus.
Priority Default = 5. Range = 1 (Lowest) to 5 (Highest)
This setting is used by ARS.
System Phone Default = None
Rights
Users set as a system phone user are able to access additional functions. The settings
are:
• None: The user cannot access any system phone options.
• Level 1: The user can access all system phone options supported on the type of phone
they are using except system management and memory card commands.
• Level 2: The user can access all system phone options supported on the type of phone
they are using including system management and memory card commands. Due to the
nature of the additional commands a login code should be set for the user to restrict
access.
Exclude From Default = Off
Directory
When on, the user does not appear in the directory list shown by the user applications
and on phones with a directory function. For users logging on as agents in an Outbound
Contact Express deployment, Exclude From Directory must be Off.
Device Type This field shows the type of phone at which the user is current logged in.
• If the user is logged out but associated with a Base Extension, the device type for the
extension port is shown.
• If the user has logged out and is not associated with a Base Extension, the device type
is listed as Device Type Unknown.

Profile Settings
Each user can be assigned to a particular profile. Each profile, other than Basic User, requires
the system to have a matching license or subscription available for the user.
The profile assigned to the user controls whether they can have a number of additional settings
enabled. The tables below list those settings and profiles. The items in ( ) brackets indicate the
default status for the settings when that profile is selected.

IP500 V2 PLDS Licensed Systems

Option Basic User Office Teleworker Mobile Power User
Worker Worker
Enable Softphone – – (On) – (On)
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User

Option Basic User Office Teleworker Mobile Power User

Worker Worker
Enable one-X Portal Services – (On) (On) – (On)
Enable one-X Telecommuter – – (On) – (On)
Enable Remote Worker [2] (Off) (Off) (On) (Off) –
Enable Desktop/Tablet VoIP – [3] (On) [3] (On) – (On)
Client
Enable Mobile VoIP Client – – – – (On)
Enable MS Teams Client – (On) (On) – (On)
Send Mobility Email – – – (Off) (Off)
Web Collaboration – (Off) (Off) – (Off)
• Not supported in IP Office R12.0
and higher.

Server Edition PLDS Licensed Systems

Option Basic User Office Worker Power User
Enable Softphone – – (On)
Enable one-X Portal Services – (On) (On)
Enable one-X Telecommuter – – (On)
Enable Remote Worker [2] (Off) (Off) (On)
Enable Desktop/Tablet VoIP [3] (Off) (On) (On)
Client
Enable Mobile VoIP Client – – (On)
Enable MS Teams Client – – (On)
Send Mobility Email – – (Off)
Web Collaboration – (Off) (Off)
• Not supported in IP Office R12.0
and higher.

Subscription Mode Systems

Option Telephony User Telephony Plus User UC User
Enable Softphone – (On) (On)
Enable one-X Portal Services – – (On)
Enable one-X Telecommuter – – (On)
Enable Remote Worker [2] (Off) (Off) (On)
Enable Desktop/Tablet VoIP – [3] (Off) (On)
Client
Enable Mobile VoIP Client – – (On)
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 User

Option Telephony User Telephony Plus User UC User

Enable MS Teams Client – – (On)
Send Mobility Email – – (Off)
Web Collaboration – – (On)
• Not supported in IP Office R12.0
and higher.

User Profile Notes:

1. Non-licensed users can be created on both Standard Mode and Server Edition systems.
2. The system supports users using remote H.323 or SIP extensions. On non-Server
Edition systems, up to 4 users are supported as remote extensions without needing to
be configured and licensed for a user profile. Additional remote users are supported if
licensed and configured for either a Teleworker or Power User user profile. On Server
Edition systems, the remote worker is supported for all user profiles.
3. Supports the Avaya Workplace Client in standalone mode only. Simultaneous mode,
shared call control mode, and presence are not available and supports local contacts only
(not enterprise or IP Office contacts). See the IP Office Avaya Workplace Client Installation
Notes manual.
• On PLDS licensed IP500 V2 systems, Avaya Workplace Client can be used with a
Basic User with IP Softphone license or a Mobile Worker with IP Softphone license.
Field Description
Profile Default = Basic User.
A user's profile controls whether they can be configured for a number of features. The
different profiles available and the features accessible by each are shown in the tables
above. The number of users that can be configured for each profile is controlled by the
user licenses or subscription that the system has.
• A Non-licensed User is allowed dial-in access, can be paged, and can be used as a
Music on Hold or Analog paging port.
• For non-subscription IP500 V2 systems, a Preferred Edition system license is a
pre-requisite for any user profile licenses.
- In a multi-site network, the Preferred Edition license of the central system is
automatically shared with other systems in the network, enabling user profile
licenses on all IP500 V2 systems.
- Each IP500 V2 system supporting a Voicemail Pro server still requires a Preferred
Edition license for Voicemail Pro operation.
• To upgrade an Office Worker or Mobile Worker to a Power User, you must first set
the user to Basic User.
• For an IP500 V2 system configured as an Avaya Cloud Office™ gateway, select the
profile ACO User. Refer to the Deploying an IP Office as an Avaya Cloud Office ATA
Gateway.
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User

Field Description
Receptionist Default = Off.
This settings allows the user to use the SoftConsole application. This requires the
configuration to have Receptionist licenses or subscriptions.
• In PLDS licensed systems, a Receptionist license is only consumed when a
configured user runs the SoftConsole application.
• In subscription systems, a Receptionist subscription is consumed when a user is
configured for SoftConsole use.
• Up to 4 users can be licensed for IP500 V2 systems, 10 for Server Edition systems.
• The use of SoftConsole is not supported for user's who then hot-desk to other systems
in a the multi-site network.
Enable Softphone Default = Controlled by the user profile, see the tables above.
If selected, the user is able to use the IP Office Softphone application.
Enable one-X Portal Default = Controlled by the user profile, see the tables above.
Services
If selected, the user is able to use the one-X Portal application, either directly or using
one of its plug-in clients.
Enable one-X Default = Controlled by the user profile, see the tables above.
Telecommuter
If selected, the user is able to use the telecommuter mode features of the one-X Portal
application. Requires Enable one-X Portal Services to also be enabled.
Enable Remote Default = Off
Worker
Indicates whether the user is allowed to use a remote H.323 or SIP extension. That is,
an extension on a different IP network from the extensions registered IP Office system.
• SIP – This option is not required for SIP extension users phones if an Avaya Session
Border Controller for Enterprise (ASBCE) is deployed in the network.
• H323 – If the user's Extension Number matches the Base Extension setting of an
IP extension, the H.323 Remote Extn Enable setting of that extension is automatically
changed to match the user's Enable Remote Worker setting and vice versa.
• Up to 4 Basic User users can be configured for Enable Remote Worker. Other users
require licensing to a profile that supports the Enable Remote Worker setting.
Enable Desktop/ Default = Controlled by the user profile, see the tables above.
Tablet VoIP Client
This option allows users to use Avaya Workplace Client on Windows or macOS
operating systems.
Enable Mobile VoIP Default = Controlled by the user profile, see the tables above.
Client
This option allows the users to use Avaya Workplace Client on Android and iOS
operating systems.
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 User

Field Description
Enable MS Teams Default = Off
Client
This option enables IP Office to fetch the Microsoft Teams user data.
The system is configured as the telephony service for calls made to and from Microsoft
Teams.
Send Mobility Email Default = Controlled by the user profile, see the tables above.
When enabled, the user receives a welcome email with the following information:
• A brief introduction of one-X Mobile Preferred for IP Office.
• Instructions and links for installing and configuring the one-X Mobile Preferred for IP
Office client.
Web Collaboration Default = Controlled by the user profile, see the tables above.
When enabled, allows the user to use the Web Collaboration application.
• Not supported in IP Office R12.0 and higher.
• In addition to the user profile license, each user requires a Web Collaboration
license.
• Web Collaboration requires Avaya one-X Portal on a Linux–based IP Office server.

User Rights
Selected user settings can be overridden by those set within a set of User Rights. The same user
rights can be applied to multiple users.
In addition, a time profile can be used to control when the user rights are applied to the user, and
whether at other times, a different set of user rights are applied or the user’s own settings.
Field Description
User Rights View This field affects Manager only. It allows you to switch between displaying the user
settings as affected by their associated Working Hours User Rights or Out of Hours
User Rights.
Working Hours Default = <None> (Continuous).
Time Profile
If set, the selected time profile defines when the user's Working Hours User Rights are
applied. Outside the time profile, the user's Out of Hours User Rights are applied
Working Hours Default = Blank (No rights restrictions).
User Rights
This field allows selection of user rights which may set and lock some user settings. If a
Working Hours Time Profile has been selected, the Working Hours User Rights are
only applied during the times defined by that time profile, otherwise they are applied at
all times.
Out of Hours User Default = Blank (No rights restrictions).
Rights
This field allows selection of alternate user rights that are used outside the times defined
by the user's Working Hours Time Profile.

Related links
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User

Voicemail
Navigation: User | Voicemail
Additional configuration information
The Enable Gmail API setting is used to configure Gmail Integration. For additional information,
see Configuring Gmail Integration on page 715.
Configuration settings
If a voicemail server application is being used on your system, each user has use of a voicemail
mailbox. You can use this form to enable this facility and various user voicemail settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.

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 Voicemail

Field Description
Voicemail Code Default = Blank. Range = 0 (no code) to 31 digits.
A code used by the voicemail server to validate access to this mailbox. If remote access is
attempted to a mailbox that has no voicemail code set, the prompt "Remote access is not
configured on this mailbox" is played.
The mailbox access code can be set through IP Office Manager or through the mailbox
telephone user interface (TUI). The minimum password length is:
• Voicemail Pro (Manager): 0
• Voicemail Pro (Intuity TUI): 2
• Embedded Voicemail (Manager): 0
• Embedded Voicemail (Intuity TUI): 0
Codes set through the Voicemail Pro telephone user interface are restricted to valid
sequences. For example, attempting to enter a code that matches the mailbox extension,
repeat the same number (111111) or a sequence of numbers (123456) are not allowed. If
these types of code are required they can be entered through Manager.
Manager does not enforce any password requirements for the code if one is set through
Manager.
• Embedded Voicemail: For Embedded Voicemail running in IP Office mailbox mode, the
voicemail code is used if set.
• IP Office mode: The voicemail code is required when accessing the mailbox from a
location that is not set as a trusted number in the user's Source Numbers list.
• Intuity Emulation mode: By default the voicemail code is required for all mailbox
access. The first time the mailbox is accessed the user will be prompted to change the
password. Also if the voicemail code setting is left blank, the caller will be prompted to
set a code when they next access the mailbox. The requirement to enter the voicemail
code can be removed by adding a customized user or default collect call flow, refer to
the Voicemail Pro manuals for full details.
• Trusted Source Access: The voicemail code is required when accessing the mailbox
from a location that is not set as a trusted number in the user's Source Numbers list.
• Call Flow Password Request: Voicemail Pro call flows containing an action where the
action's PIN code set to $ will prompt the user for their voicemail code.
• Changing the Code: All of the voicemail interfaces, except IMS and IMAP, provide
options for the user to change the voicemail code themselves. In addition, Voicemail Pro
running in Intuity emulation mode will request that the user sets a code when they first
log in to their mailbox using the phone.
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User

Field Description
Voicemail On Default = On.
When on, the mailbox is used by the system to answer the user's unanswered calls or
calls when the user's extension returns busy. Note that selecting off does not disable use
of the user's mailbox. Messages can still be forward to their mailbox and recordings can
be placed in it. The mailbox can also still be accessed to collect messages.
When a caller is directed to voicemail to leave a message, the system indicates the target
user or hunt group mailbox.
• The mailbox of the originally targeted user or hunt group is used. This applies even if the
call has been forwarded to another destination. It also includes scenarios where a hunt
group call overflows or is in fallback to another group.
• Voicemail Pro can be used to customize which mailbox is used separately from the
mailbox indicated by the system.
Voicemail Help Default = Off
This option controls whether users retrieving messages are automatically given an
additional prompt "For help at any time press 8." If switched off, users can still press
8 for help. For voicemail systems running in Intuity emulation mode, this option has no
effect. On those systems the default access greeting always includes the prompt "For help
at any time, press *4" (*H in the US locale).
Voicemail Default = Off
Ringback
When enabled and a new message has been received, the voicemail server calls the
user's extension to attempt to deliver the message each time the telephone is put down.
Voicemail will not ring the extension more than once every 30 seconds.
Voicemail Email Default = Off
Reading
This option can be enabled for users whose Profile is set to Mobile Worker or Power
User. If enabled, when you log into you voicemail box, it will detect your email messages
and read them to you. This email text to speech feature is set-up through Voicemail Pro.
This option is not currently supported with Linux based Voicemail Pro.
UMS Web Default = On
Services
When selected, the user can use any of the Voicemail Pro UMS services to access their
voicemail messages (IMAP email client, web browser or Exchange 2007 mailbox). Note
that the user must have a voicemail code set in order to use the UMS services.
• For subscription systems, this setting is only supported for UC User.
• For PLDS licensed systems, this setting is only supported for Teleworker, Office
Worker or Power User users.
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 Voicemail

Field Description
Enable Gmail API Default = Off.
This setting is only supported on Server Edition systems and requires the used to have
UMS Web Services enabled. When enabled:
• The Voicemail Email setting is disabled.
• The Voicemail Email Mode options (Off, Copy, Forward, Alert) are available.
This feature uses the Gmail address defined in the setting User | User | Unique Identity.
Voicemail Email Default = Blank (No voicemail email features)
This field is used to set the user or group email address used by the voicemail server
for voicemail email operation. When an address is entered, the additional Voicemail Email
control below are selectable to configure the type of voicemail email service that should be
provided.
Use of voicemail email requires the Voicemail Pro server to have been configured to
use either a local MAPI email client or an SMTP email server account. For Embedded
Voicemail, voicemail email is supported and uses the system's SMTP settings.
The use of voicemail email for the sending (automatic or manual) of email messages
with wav files attached should be considered with care. A one-minute message creates a
1MB .wav file. Many email systems impose limits on emails and email attachment sizes.
For example the default limit on an Exchange server is 5MB.

Note:
Unicode characters are not supported.
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User

Field Description
Voicemail Email Default = Off
Mode
This option is selectable when for users and groups when either:
• A Voicemail Email email address is set.
• The Enable Gmail API is set to On.
These settings control the mode of automatic voicemail email operation provided by the
voicemail server whenever the voicemail mailbox receives a new voicemail message.
Users can change their voicemail email mode using visual voice. The ability to change the
voicemail email mode can also be provided by Voicemail Pro in a call flow using a Play
Configuration Menu action or a Generic action.
If the voicemail server is set to IP Office mode
• Users can change their voicemail email mode through the telephone prompts.
• users can manually forward a message to email.
The options are:
• Off If off, none of the options below are used for automatic voicemail email. Users can
also select this mode by dialing *03 from their extension.
• Copy If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a copy of the message is attached to an email and sent to the
email address. There is no mailbox synchronization between the email and voicemail
mailboxes. For example reading and deletion of the email message does not affect the
message in the voicemail mailbox or the message waiting indication provided for that
new message.
• Forward If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, that message is attached to an email and sent to the email address.
No copy of the voicemail message is retained in the voicemail mailbox and their is no
message waiting indication. As with Copy, there is no mailbox synchronization between
the email and voicemail mailboxes. Users can also select this mode by dialing *01 from
their extension.
Note that until email forwarding is completed, the message is present in the voicemail
server mailbox and so may trigger features such as message waiting indication.
• UMS Exchange 2007 With Voicemail Pro, the system supports voicemail email to an
Exchange 2007 server email account. For users and groups also enabled for UMS
Web Services this significantly changes their mailbox operation. The Exchange Server
inbox is used as their voicemail message store and features such as message waiting
indication are set by new messages in that location rather than the voicemail mailbox on
the voicemail server. Telephone access to voicemail messages, including Visual Voice
access, is redirected to the Exchange 2007 mailbox.
• Alert If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a simple email message is sent to the email address. This is an email
message announcing details of the voicemail message but with no copy of the voicemail
message attached. Users can also select this mode by dialing *02 from their extension.
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 Voicemail

Field Description
DTMF Breakout
When a caller is directed to voicemail to leave a message, they can be given the option to be transferred to
a different extension. The greeting message needs to be recorded telling the caller the options available. The
extension numbers that they can be transferred to are entered in the fields below.System default values can be
set for these numbers and are used unless a different number is set within these user settings. The values can
be set using User Rights.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for Enterprise Branch with CS 1000 deployments) is configured as the central
voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant operation. The
Park & Page feature allows a call to be parked while a page is made to a hunt group or extension. This feature
can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
Reception/ The number to which a caller is transferred if they press 0while listening to the mailbox
Breakout (DTMF greeting rather than leaving a message (*0 on Embedded Voicemail in IP Office mode).
0)
For voicemail systems set to Intuity emulation mode, the mailbox owner can also access
this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave Mail
action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success results
connections depending on whether the caller pressed 0 before or after the record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0) setting.
When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
• Paging Number – displays a list of hunt groups and users (extensions). Select a hunt
group or extension to configure this option.
• Retries – the range is 0 to 5. The default setting is 0.
• Retry Timeout – provided in the format M:SS (minute:seconds). The range can be set
in 15-second increments. The minimum setting is 15 seconds and the maximum setting
is 5 minutes. The default setting is 15 seconds
Breakout (DTMF The number to which a caller is transferred if they press 2while listening to the mailbox
2) greeting rather than leaving a message (*2 on Embedded Voicemail in IP Office mode).
Breakout (DTMF The number to which a caller is transferred if they press 3while listening to the mailbox
3) greeting rather than leaving a message (*3 on Embedded Voicemail in IP Office mode).

Related links
User on page 433

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User

DND
Navigation: User | DND
Do not disturb prevents the user from receiving hunt group and page calls. Direct callers hear
busy tone or are diverted to voicemail if available. It overrides any call forwarding, follow me and
call coverage settings. A set of exception numbers can be added to list numbers from which the
user still wants to be able to receive calls when they have do not disturb in use.
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Do Not Disturb Default = Off
When checked the user's extension is considered busy, except for calls coming from
sources listed in their Do Not Disturb Exception List. When a user has do not disturb in
use, their normal extension will give alternate dialtone when off hook. Users with DND on
are indicated as 'busy' on any BLF indicators set to that user.
Do Not Disturb Default = Blank
Exception List
This is the list of telephone numbers that are still allowed through when Do Not Disturb
is set. For example this could be an assistant or an expected phone call. Internal
extension numbers or external telephone numbers can be entered. If you wish to add
a range of numbers, you can either enter each number separately or make use of the
wildcards "N" and "X" in the number. For example, to allow all numbers from 7325551000
to 7325551099, the DND Exception number can be entered as either 73255510XX or
73255510N. Note that this list is only applied to direct calls to the user.
Calls to a hunt group of which the user is a member do not use the Do Not Disturb
Exceptions list.

Related links
User on page 433

Short Codes
Navigation: User | Short Codes
Additional configuration information
For additional configuration information on short codes, see Short Code Overview on page 933.
Configuration settings
Short codes entered in this list can only be dialed by the user. They will override any matching
user rights or system short code.

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 Source Numbers

User and User Rights short codes are only applied to numbers dialed by that user. For example
they are not applied to calls forwarded via the user.
Warning:
User dialing of emergency numbers must not be blocked by the addition of short codes.
If short codes are added, the users ability to dial emergency numbers must be tested and
maintained.
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Code Description
*FWD Short codes of this form are inserted by the system. They are used in conjunction
with the User | Forwarding settings to remember previously used forwarding
numbers. They can be accessed on that tab by using the drop-down selector on
the forwarding fields.
*DCP Short codes of this form are often inserted by the system. They are used by some
phone types to contain settings relating to functions such as ring volume and auto
answer. Deleting such short codes will cause related phone settings to return to
their defaults.
*DCP/Dial/ For systems with TCM phone ports, when a phone is first connected to the port,
8xxxxxxx,0,1,1,0/0 the button programming of the associated user is overwritten with the default
button programming appropriate for the phone model. Adding the above short
code prevents that behavior if not required, for example if a pre-built configuration
including user button programming is added to the system before the connection of
phones.

Related links
User on page 433

Source Numbers
Navigation: User | Source Numbers
Source numbers are used to configure features which do not have specific controls within the
IP Office Manager or IP Office Web Manager interfaces. For more details, see User Source
Numbers on page 795.
Sources numbers are divided into two types:
• User source numbers are used to apply settings to individual users.
• NoUser source numbers are used to apply settings to the IP Office system or to all users on
the system.

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User

Related links
User on page 433

Telephony
Navigation: User | Telephony
This form allows you to set telephony related features for the user. These override any matching
setting in the System | Telephony tab. The settings are grouped into a number of sub-tabs.
Related links
User on page 433
Call Settings on page 450
Supervisor Settings on page 453
Multi-line Options on page 457
Call Log on page 459
TUI on page 460

Call Settings
Navigation: User | Telephony | Call Settings
Additional configuration information
For additional information on ring tones, see Ring Tones on page 655.
Configuration settings
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Outside Call Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for external calls to the user.
The distinctive ring patterns used for other phones are fixed. Note that changing the
pattern for users associated with fax and modem device extensions can cause those
devices to not recognize and answer calls.
Inside Call Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for internal calls to the user.
The distinctive ring patterns used for other phones are fixed.
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Field Description
Ring Back Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for ringback calls to the user.
The distinctive ring patterns used for other phones are fixed.
No Answer Time Default = Blank (Use system setting). Range = 6 to 99999 seconds.
Sets how long a call rings the user before following forwarded on no answer if set or
going to voicemail. Leave blank to use the system default setting (System > Telephony >
Telephony > Default No Answer Time).
• For users who are using Avaya Workplace Client on iOS devices, it is recommended to
set the time to at least 20 seconds.
Wrap-up Time Default = 2 seconds, Range 0 to 99999 seconds. Specifies the amount of time after
(secs) ending one call during which the user is treated as still being busy. During this time:
• Other phones or applications monitoring the user's status indicate the user as still being
busy (on a call).
• Hunt group calls are not presented to the user.
• If the user is using a single line set, direct calls also receive busy treatment. If the
user is using a mutli-line set (multiple call appearances), direct calls to them will ring as
normal.
• It is recommended that this option is not set to less than the default of 2 seconds. 0 is
used to allow immediate ringing.
• The user's wrap-up time setting is added to the system hold recall time for calls put on
hold by the user.
• For users set as a CCR Agent, use the setting User | Telephony | Supervisor
Settings | After Call Work Time.
Transfer Return Default = Blank (Off), Range 1 to 99999 seconds.
Time (secs)
Sets the delay after which any call transferred by the user, which remains unanswered,
should return to the user. A return call will continue ringing and does not follow any
forwards or go to voicemail.
• Transfer return will occur if the user has an available call appearance button.
• Transfer return is not applied if the transfer is to a hunt group that has queuing enabled.
Call Cost Mark-Up Default = 100.
This setting is used for ISDN advice of charge (AOC). The markup is applied to the cost
calculations based on the number of units and the line base cost per charging unit. The
field is in units of 1/100th, for example an entry of 100 is a markup factor of 1. This value
is included in the system SMDR output.
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User

Field Description
Advertize Callee Default = System Default (Off).
State To Internal
The options are:
Callers
• System Default (Off). The system setting is System | Telephony | Telephony |
Advertize Callee State To Internal Callers.
• On
• Off
When enabled, for internal calls, additional status information is communicated to the
calling party.
Not supported for SIP endpoints except the J100 Series (excluding the J129).
• When calling another internal phone and the called phone is set to Do Not Disturb or
on another call, the calling phone displays “Do Not Disturb” or “On Another Call” rather
than “Number Busy”.
• On 9500 Series, 9600 Series and J100 Series phones, if a line appearance is
programmed on a button on phone A and that line is in use on phone B, then phone A
displays the name of the current user of the line along with the line number.
• If a line appearance on a phone is in use elsewhere in the system and
another extension unsuccessfully attempts to seize that line, the phone displays “In
Use:<name>” where <name> is the name of the user currently using the line.
Call Waiting On Default = Off
For users on phones without appearance buttons, if the user is on a call and a second
call arrives for them, an audio tone can be given in the speech path to indicate a waiting
call (the call waiting tone varies according to locale). The waiting caller hears ringing
rather than receiving busy. There can only be one waiting call, any further calls receive
normal busy treatment. If the call waiting is not answered within the no answer time, it
follows forward on no answer or goes to voicemail as appropriate. User call waiting is not
used for users on phones with multiple call appearance buttons.
Answer Call Default = On
Waiting on Hold
Applies to analog and IP DECT extension users only. If the user has a call waiting and
places their current call on hold, the waiting call is automatically connected.
Busy on Held Default = Off for users with call appearance buttons/On for other users.
If on, when the user has a call on hold, new calls receive busy treatment. They will
follow the user's forward on busy setting or are diverted to voicemail. Otherwise busy tone
(ringing for incoming analog calls) is played. This overrides call waiting when the user has
a call on hold. The use of Busy on Held for users with multiple call appearance buttons is
deprecated and Manager will prompt whether it should switch off the feature off for such a
user.
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 Telephony

Field Description
Offhook Station Default = Off
Off-hook station allows an analog extension to be left permanently off-hook, with calls
being made and answered using an application or TAPI. When enabled, the analog
extension user is able to control calls using the application in the following ways:
Offhook station does not disable the physical off-hook on the phone. When starting
with the phone on-hook, making and answering calls is the same as normal analog
extension operation. Additionally however calls can be initiated from the application. After
entering the required number and making the call, the on-hook analog extension receives
a ringback showing the users own caller ID and when answered the outgoing call leg to
the dialed number is started. Calls to a busy destination present busy tone before being
cleared.
The application can be used to end a call with the analog extension still off-hook. Instead
of hearing disconnect tone the user hears silence and can use the application to make
another call. Though off-hook the user is indicated as idle on BLF indicators. Without
off-hook Station set the user would be indicated as busy when off-hook, whether on a call
or not.
If off-hook and idle (having cleared a previous call), incoming call alerts by presenting
ringing through the audio path. The call can be answered using the application or going
on-hook/off-hook or by pressing recall. Note that if the phone normally displays call ID,
any caller ID displayed on the phone is not updated in this mode, however the call ID in
the application will be that of the current call.
If on-hook, an incoming call alerts as normal using the phone's ringer and is answered by
going off-hook. The answer call option in the application cannot be used to answer calls
to an on-hook analog extension.
While off-hook and idle, the analog extension user will receive page calls.
If the analog extension handset is replaced with a headset, changing the Manager setting
Extension | Analog | Equipment Classification to Quiet Handset is recommended.

Related links
Telephony on page 450

Supervisor Settings
Navigation: User | Telephony | Supervisor Settings
Additional configuration information
• For additional information on the Force Authorization Code setting, see Configuring
Authorization Codes on page 705.
• For additional information on the Inhibit Off-Switch Forward/Transfers see, Off-Switch
Transfer Restrictions on page 784.
Configuration settings
These settings relate to user features normally only adjusted by the user's supervisor.

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User

The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Login Code Default = Blank. Range = Up to 31 digits.
• Login code must be at least 4 digits for DS port users.
• Login codes of up to 15 digits are supported with Extn Login buttons.
• Login codes of up to 31 digits are supported with Extn Login short codes.
This code is used for logging in on a phone (and for restricting access to features on
phones. See Hot Desking on page 758.
• Hot desking is not supported for centralized users. Centralized users use the Login
Code for SIP registration on Session Manager.
• Normally users can only log out if they have a Login Code set or if they are currently
logged in at an extension whose Base Extension number no longer matches their own
Extension setting.
• When set, the short code feature Change Login Code can be used by users to change
their login code.
• If the user has a login code set, it is used by the Outgoing Call Bar Off short code
feature.
• If the user has a login code set, access to a range of programmable button features
requires entry of the login code. For example, access Self Admin and System Phone
features.
Login Idle Period Default = Blank (Off). Range = 0 (Off) to 99999.
(secs)
If the telephone is not used for this period; the user currently logged in is automatically
logged out. This option should be used only in conjunction with Force Login (see below).
Monitor Group Default = <None>
Sets the hunt group whose members the user can monitor if silent monitoring is setup.
See the Call Listen short code.
Privacy Override Default = <None>
Group
The drop-down menu lists the local and network advertised hunt groups. If selected, calls
to this user cannot be seen or picked up by other users unless they are a member of the
selected group.
Coverage Group Default = <None>.
If a group is selected, then in scenarios where an external call would normally have
gone to voicemail, it instead continues ringing and also starts alerting the members of the
coverage group. See Coverage Groups on page 775.
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 Telephony

Field Description
Status on No Default = Logged On.
Answer
Hunt groups can change the status of call center agents (users with a log in code and
set to forced log in) who do not answer a hunt group call presented to them before it is
automatically presented to the next agent. Use of this is controlled by the Agent's Status
on No Answer Applies To setting of the hunt group. This option is not used for calls
ringing the agent because the agent is in another group's overflow group. The options are:
• Logged On: If this option is selected, the user's status is not changed.
• Busy Wrap-Up: If this option is selected the user's membership status of the hunt group
triggering the action is changed to disabled. The user can still make and receive calls
and will still continue to receive calls from other hunt groups to which they belong.
• Busy Not Available: If this option is selected the user's status is changed to do not
disturb. This is the equivalent of DND and will affect all calls to the user.
• Logged Off: If this option is selected the users status is changed to logged out. In that
state they cannot make calls or receive calls. Hunt group calls go to the next available
agent and personal calls treat the user as being busy.
Reset Longest Default = All Calls.
Idle Time
This setting is used in conjunction with hunt groups set to Longest Waiting (also known as
Idle and Longest Waiting). It defines what type of calls reset the idle time of users who are
members of these hunt groups. Options are All Calls and External Incoming.
ICR Agent Role Note:
This field is available only if you first configure the user as an Integrated Contact
Reporter (ICR) user using the ICR Agent field, which is provided near the end.
Default = Agent.
Select Supervisor to make the user a supervisor. Selecting Supervisor displays the
Enable Huntgroup Monitoring area and lists all the hunt groups available for the
supervisor to monitor. The hunt groups are listed only if they were already configured.
Select the hunt groups for supervisor to monitor.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Force Login Default = Off
If checked, the user must log in using their Login Code to use any extension including an
extension to which they are the default associated user (Base Extension).
For example: If user B has logged onto user A's phone and now logs off
• If user A has Force Login enabled, they are not automatically logged back on to their
extension.
• If user A do not have Force Login enabled, they are automatically logged back in.
Force Account Default = Off
Code
If checked, the user must enter a valid account code to make an external call.
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User

Field Description
Force Default = Off.
Authorization
If checked, the user must enter a valid authorization code to make an external call. That
Code
authorization code must be one associated with the user or the user rights to which the
user belongs.
Incoming Call Bar Default = Off
When enabled, this setting stops a user from receiving any external calls. On the calling
phone, the call is rejected.
Outgoing Call Bar Default = Off
When enabled, this setting stops a user from making any external calls except those that
use dial emergency features. On many Avaya display phones, this causes a B to be
displayed. The following features can be used with outgoing call bar: Outgoing Call Bar
On, Outgoing Call Bar Off and Change Login Code.
Inhibit Off-Switch Default = Off.
Forward/Transfers
When enabled, this setting stops the user from transferring or forwarding calls externally.
This does not stop another user transferring the restricted users calls off-switch on their
behalf. Note that a number of other controls may inhibit the transfer operation.
Can Intrude Default = Off
If enabled, the user can perform is allowed to perform a range of action on other user's
calls. For example: Call Intrude, Call Listen, Call Steal and Dial Inclusion. See Call
Intrusion on page 716.
• Use of the features is subject to the Cannot Be Intruded setting of the target.
Cannot be Default = On
Intruded
If checked, this user's calls cannot be interrupted or acquired by users who have Can
Intrude enabled. This setting also affects whether other users can use their appearance
buttons to bridge into a call to which this user has been the longest present user.
Can Trace Calls Default = Off. This settings controls whether the user is able to make used of ISDN MCID
controls.
ICR Agent Default = Off.
Enable to make the user an ICR user. If enabled, the ICR Agent Role field becomes
available and the After Call Work related fields are activated.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
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 Telephony

Field Description
Automatic After Default = Off.
Call Work
If enabled, the agent goes into After Call Work (ACW) at the end of an ICR and non-ICR
hunt group call to indicate that they are busy with post-call processing activity. During the
ACW state, they are not sent any hunt group calls.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Can Control After Default = Off.
Call Work
If enabled, the agent can extend the currently active After Call Work time indefinitely.
After Call Work Default = The value in this field is populated from the Default After Call Work Time field
Time (Sec) located at System | Contact Center.
The time after a call when an agent is busy and unable to deal with hunt group calls.
Change the value if you want to specify ACW time for this user to be different from the
system default.
Can Accept Default = Off [Brazil Only]
Collect Calls
Determines whether the user is able to receive and accept collect calls.
Deny Auto Default = Off.
Intercom Calls
When enabled, any automatic intercom calls to the user's extension are automatically
turned into normal calls.
Enable Hunt Default = Blank
group Monitoring
All the available hunt groups for Integrated Contact Reporter are listed under Hunt Group
Name. Select the check box against the hunt group to enable it for monitoring by the
supervisor. Select the Hunt Group Name check box to enable all the hunt groups for
monitoring by the supervisor. The field is activated if you assign the user with Supervisor
role using the ICR Agent Role field.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Related links
Telephony on page 450

Multi-line Options
Navigation: User | Telephony | Multi-line Options
Additional configuration information
• For additional configuration information, see Appearance Button Operation on page 1159.
• For the Reserve Last CA setting, 1400, 1600, 9500 and 9600 Series telephone users
can put a call on hold pending transfer if they already have held calls even if they have
no free call appearance button available. For additional information, see Context Sensitive
Transfer on page 785.

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User

Configuration settings
Multi-line options are applied to a user's phone when the user is using an Avaya phones which
supports appearance buttons (call appearance, line appearance, bridged and call coverage).
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Individual Default = 10 seconds, Range 1 to 99999 seconds.
Coverage Time
(secs) This function sets how long the phone will ring at your extension before also alerting at
any call coverage users. This time setting should not be equal to or greater than the No
Answer Time applicable for the user.
Ring Delay Default = Blank (Use system setting). Range = 0 (use system setting) to 98 seconds.
This setting is used when any of the user's programmed appearance buttons is set to
Delayed ringing. Calls received on that button will initially only alert visually. Audible
alerting will only occur after the ring delay has expired.
Coverage Ring Default = Ring.
This field selects the type of ringing that should be used for calls alerting on any the user's
call coverage and bridged appearance buttons. Ring selects normal ringing. Abbreviated
Ring selects a single non-repeated ring. No Ring disables audible ringing. Note that each
button's own ring settings (Immediate, Delayed Ring or No Ring) are still applied.
The ring used for a call alerting on a call coverage or bridged appearance button will vary
according to whether the user is currently connected to a call or not.
• If not currently on a call, the Coverage Ring setting is used.
• If currently on a call, the quieter of the Coverage Ring and Attention Ring settings is
used.

Attention Ring Coverage Ring Setting

Setting Ring Abbreviated Off
Ring Ring Abbreviated Off
Abbreviated Abbreviated Abbreviated Off
Attention Ring Default = Abbreviated Ring.
This field selects the type of ringing that should be used for calls alerting on appearance
buttons when the user already has a connected call on one of their appearance buttons.
• Ring selects normal ringing.
• Abbreviated Ring selects a single ring.
• Note that each buttons own ring settings (Immediate, Delayed Ring or No Ring) are still
applied.
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 Telephony

Field Description
Ringing Line Default = On.
Preference
For users with multiple appearance buttons. When the user is free and has several
calls alerting, ringing line preference assigns currently selected button status to the
appearance button of the longest waiting call. Ringing Line Preference overrides Idle
Line Preference.
Idle Line Default = On.
Preference
For users with multiple appearance buttons. When the user is free and has no alerting
calls, idle line preference assigns the currently selected button status to the first available
appearance button.
Delayed Ring Default = Off.
Preference
This setting is used in conjunction with appearance buttons set to delayed or no ring. It sets
whether ringing line preference should use or ignore the delayed ring settings applied to
the user's appearance buttons.
• When on, ringing line preference is only applied to alerting buttons on which the ring
delay has expired.
• When off, ringing line preference can be applied to an alerting button even if it has
delayed ring applied.
Answer Pre- Default = Off.
Select
Normally when a user has multiple alerting calls, only the details and functions for the call
on currently selected button are shown. Pressing any of the alerting buttons will answer the
call on that button, going off-hook will answer the currently selected button.
Enabling Answer Pre-Select allows the user to press any alerting button to make it the
current selected button and displaying its call details without answering that call until the
user either presses that button again or goes off-hook.
Note that when both Answer Pre-Select and Ringing Line Preference are enabled, once
current selected status is assigned to a button through ringing line preference, it is not
automatically moved to any other button.
Reserve Last CA Default = Off.
When selected, this option stops the user's last call appearance button from being used
to receive incoming calls. This ensures that the user always has a call appearance
button available to make an outgoing call and to initiate actions such as transfers and
conferences.
• 1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending
transfer if they already have held calls even if they have no free call appearance button
available.

Related links
Telephony on page 450

Call Log
Navigation: User | Telephony | Call Log

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User

The IP Office stores a centralized call log for each user:

• The user's centralized call log is stored by the IP Office system on which the user is
configured. If the user is logged in on another system, new call log records are sent to
the user's home IP Office system, but using the time and date on the IP Office system where
the user is logged in.
• Each user's centralized call log can contain 30 (IP500 V2) or 60 (Server Edition) call records.
Each new call record replaces the oldest previous record when it reaches the limit.
• By default, the centralized call log is displayed on Avaya phones with a fixed Call Log
or History button (1400, 1600, 9500, 9600, J100 Series) and in the one-X Portal, Avaya
Workplace Client, and IP Office User Portal applications.
• The centralized call log moves with the user as they log onto different phones and IP Office
applications.
• The missed call count is updated per unique caller, not per call.
These settings are using in conjunction with the system wide call log settings (System >
Telephony > Call Log).
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Centralized Call Default = System Default (On)
Log
This setting allows the use of centralized call logging to be enabled or disabled on a per
user basis. The default is to match the system setting System | Telephony | Call Log |
Default Centralized Call Log On.
The other options are On or Off for the individual user. If set to Off, the user receives the
message “Call Log Disabled” when the Call Log button is pressed.
Delete Default = 00:00 (Never).
records after
(hours:minutes) If a time period is set, records in the user's call log are automatically deleted after this
period.
Groups Default = System Default (On).
This section contains a list of hunt groups on the system. If System | Telephony | Call
Log | Log Missed Huntgroup Calls is enabled, then the selected groups are shown as
part of the users centralized call log.
The missed hunt group calls are any missed calls for the whole hunt group, not just group
calls presented to the user and not answered by them.

Related links
Telephony on page 450

TUI
Navigation: User | Telephony | TUI

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 Forwarding

Used to configure system wide telephony user interface (TUI) options for 1400, 1600, 9500, 9600
and J100 Series phones (except the J129).
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Features Menu Controls
User Setting Default = Same as System
When set to Same as System, matches the system-wide settings of the System
| Telephony | TUI menu options. When set to Custom, uses the Features Menu
settings below.
Features Menu Default = On
When set to off, TUI feature menus are not available. When set to on, you can select
to turn individual feature menus off or on. The following feature menus are listed:
• Basic Call Functions: If selected, users can access menu options for call pickup,
park, unpark and transfer to mobile functions.
• Advanced Call Functions: If selected, users can access the menu options for
do not disturb, account code, withhold number and internal auto-answer functions.
Note, the Account Code menu is only shown if the system has been configured
with accounts codes.
• Forwarding: If selected, users can access the phone's menus for forwarding and
follow me functions.
• Hot Desk Functions: If selected, users can access the menu options for logging in
and out.
• Passcode Change: If selected, users can change their login code (security
credentials) through the phone menus..
• Phone Lock: If selected, users can access the menu options for locking the phone
and for setting it to automatically lock.
• Self Administration: If selected, users can access the phone’s Self-
Administration menu options.
• Voicemail Controls: If set, users can access the Visual Voice option through the
phone's Features menu.

Related links
Telephony on page 450

Forwarding
Navigation: User | Forwarding
Additional configuration information
For additional configuration information, see DND, Follow Me, and Forwarding on page 744.

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User

Configuration settings
Use this page to check and adjust a user's call forwarding and follow me settings. For additional
configuration information, see DND, Follow Me, and Forwarding on page 744.
Follow Me is intended for use when the user is present to answer calls but for some reason is
working at another extension. For example; temporarily sitting at a colleague's desk or in another
office or meeting room. As a user, you would use Follow Me instead of Hot-Desking if you do not
have a log in code or you do not want to interrupt you colleague also receiving their own calls.
Multiple users can use follow me to the same phone.
Forwarding is intended for use when, for some reason, the user is unable to answer a call. They
may be busy on other calls, unavailable or simply do not answer. Calls may be forwarded to
internal or, subject to the user's call barring controls, external numbers.
• To bar a user from forwarding calls to an external number: Select User | Telephony |
Supervisor Settings | Inhibit Off-Switch Forward/Transfers.
• To bar all users from forwarding calls to external numbers: Select System | Telephony |
Telephony | Inhibit Off-Switch Forward/Transfers.
Note that analog lines doe not provide call progress signalling. Therefore, calls forwarded off-
switch via an analog line are treated as answered and are not recalled.
Once a call has been forwarded to an internal target, it will ignore the target’s Forward No
Answer or Forward on Busy settings but may use its Forward Unconditional settings unless
they create a loop.
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
General Settings
Field Description
Block Forwarding Default = Off.

When enabled, call forwarding is blocked for this user. The following actions are blocked:
Follow me, Forward unconditional, Forward on busy, Forward on no answer and Hot
Desking.
Follow Me Default = Blank. Range = Internal extension number.
Number
Redirects the user's calls to the internal extension number entered. If the redirected call
receives busy or is not answered, it follows the user's forwarding and or voicemail settings
as if it had been presented to their normal extension. When a user has follow me in
use, their normal extension will give alternate dialtone when off hook. Using Follow Me
overrides Forward Unconditional.
Calls targeting longest waiting type hunt groups ignore Follow Me.
Calls triggered by actions at the user's original extension, for example voicemail ringback,
ignore Follow Me.
Park, hold and transfer return calls will go to the extension at which the user initiated the
park, hold or transfer action.

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 Forwarding

Forward Unconditional
Field Description
Forward Default = Off
Unconditional
This option, when checked and a Forward Number also set, forwards all external calls
immediately. Additional options allow this forwarding to also be applied to internal calls and
to hunt group calls if required. When a user has forward unconditional in use, their normal
extension will give alternate dialtone when off hook. If the destination is an internal user on
the same system, they are able to transfer calls back to the user, overriding the Forward
Unconditional.
After being forwarded for the user’s no answer time, if still unanswered, the system can
apply additional options. It does this if the user has forward on no answer set for the call
type or if the user has voicemail enabled.
• If the user has forward on no answer set for the call type, the call is recalled and then
forwarded to the forward on no answer destination.
• If the user has voicemail enabled, the call is redirected to voicemail.
• If the user has both options set, the call is recalled and then forwarded to the forward on
no answer destination for their no answer time and then if still unanswered, redirected to
voicemail.
• If the user has neither option set, the call remains redirected by the forward unconditional
settings.
Note that for calls redirected via external trunks, detecting if the call is still unanswered
requires call progress indication. For example, analog lines do not provide call progress
signalling and therefore calls forwarded via an analog lines are treated as answered and
not recalled.
To Voicemail Default = Off.
If selected and forward unconditional is enabled, calls are forwarded to the user's
voicemail mailbox. The Forward Number and Forward Hunt Group Calls settings are
not used. This option is not available if the system's Voicemail Type is set to None. 1400,
1600, 9500 and 9600 Series phone users can select this setting through the phone menu.
Note that if the user disables forward unconditional the To Voicemail setting is cleared.
Forward Number Default = Blank. Range = Internal or External number. Up to 33 characters.
This option sets the destination number to which calls are forwarded when Forward
Unconditional is checked. The number can be an internal or external number. This option
is also used for Forward on Busy and Forward on No Answer if no separate Forward
Number is set for those features. If a user forwards a call to a hunt group of which they are
a member, the group call is not presented to them but is presented to other members of
the hunt group.
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User

Field Description
Forward Hunt Default = Off
Group Calls
Hunt group calls (internal and external) are not normally presented to a user who has
forward unconditional active. Instead they are presented to the next available member
of the hunt group. This option, when checked, sets that hunt group calls (internal and
external) are also forwarded when forward unconditional is active. The group's Ring Type
must be Sequential or Rotary, not Collective or Longest Waiting. The call is forwarded
for the period defined by the hunt group's No Answer Time after which it returns to the
hunt group if unanswered. Note also that hunt group calls cannot be forwarded to another
hunt group.
Forward Internal Default = On.
Calls
This option, when checked, sets that internal calls should be also be forwarded
immediately when forward unconditional is active.

Forward on Busy/No Answer

Field Description
Forward On Busy Default = Off
When checked and a forward number is set, external calls are forwarded when the user's
extension is busy. The number used is either the Forward Number set for Forward
Unconditional or if set, the separate Forward Number set under Forward On Busy.
Having Forward Unconditional active overrides Forward on Busy.
If the user has Busy on Held selected, if forward on busy is active it is applied when the
user is free to receive calls but already has a call on hold.
If the user's phone has multiple call appearance buttons, the system will not treat them as
busy until all the call appearance buttons are in use unless the last appearance button has
been reserved for outgoing calls only.
Forward On No Default = Off When checked and a forward number is set, calls are forwarded when the
Answer user does not answer within their set No Answer Time (User | Telephony | Call Settings).
Forward Number Default = Blank. Range = Internal or External number. Up to 33 characters.
If set, this number is used as the destination for Forward On Busy and Forward On
No Answer when on. If not set, the Forward Number set for Forward Unconditional is
used. If a user forwards a call to a hunt group of which they are a member, the group call is
not presented to them but is presented to other members of the hunt group.
Forward Internal Default = On. When checked, this option sets that internal calls should be also be
Calls forwarded when forward on no answer or forward on busy is active.

Related links
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 Dial In

Dial In
Navigation: User | Dial In
Use this dialogue box to enable dial in access for a remote user. An Incoming Call Route and RAS
service must also be configured.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Dial In On Default = Off
When enabled, dial in access into the system is available via this user.
Dial In Time Default = <None>
Profile
Select the Time Profile applicable to this User account. A Time Profile can be used to set
time restrictions on dial in access via this User account. Dial In is allowed during the times
set in the Time Profile form. If left blank, then there are no restrictions.
Dial In Firewall Default = <None>
Profile
Select the Firewall Profile to restrict access to the system via this User account. If blank,
there are no Dial In restrictions.

Related links
User on page 433

Voice Recording
Navigation: User | Voice Recording
These settings are used to control manual and automatic recording of user's calls.
• Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
• Call recording starts when the call is answered.
• Call recording is paused when the call is parked or held. It restarts when the call is unparked
or taken off hold. This does not apply to SIP terminals.
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Recording continues for the duration of the call or up to the maximum recording time
configured on the voicemail server.
• Recording is stopped when the call ends or if:
- User call recording stops if the call is transferred to another user.
- Account code call recording stops if the call is transferred to another user.
- Hunt group call recording stops if the call is transferred to another user who is not a
member of the hunt group.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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User

Related links
User on page 433

Button Programming
Navigation: User | Button Programming
Additional configuration information
For additional information on programming button actions, see Button Programming Overview on
page 1042.
For a description of each button action, see Button Programming Actions on page 1046.
Used to assign functions to the programmable keys provided on many Avaya telephones. For full
details of button programming refer to the section Button Programming.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Button No. The number of the DSS key against which the function is being set. To set a function
against a button double-click it or select it and then click Edit.
Label This is a text label for display on the phone. If no label is entered, the default label for the
selected action is used.
Action Defines the action taken by the menu item.
Action Data This is a parameter used by the selected action. The options here will vary according to
the selected button action.
Display All The number of button displayed is based on the phone associated with the user when the
configuration was loaded. This can be overridden by selecting Display All Buttons. This
may be necessary for users who switch between different phones using hot desking or
have an expansion unit attached to their phone.

Related links
User on page 433

Menu Programming
Navigation: User | Menu Programming
These menus control a range of options that are specific to different types of phones. The
functions become accessible when the user logs in on the appropriate type of phone.
Related links
User on page 433
Huntgroup on page 467

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 Menu Programming

4400/6400 on page 467

Huntgroup
Navigation: User | Menu Programming | Hunt Group
Avaya 1400, 1600, 9500 and 9600 Series phone users can control various settings for selected
hunt groups. These settings are also used for one-X Portal for IP Office.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Can Change Default = Off
Membership
This list shows the hunt groups of which the user is a member. Up to 10 of these groups
can be checked; those group and the users current membership status are then displayed
on the phone. The user can change their membership status through the phone's menus.
Can Change Default = Off
Service Status
This list shows all the hunt groups on the system. Up to 10 of these groups can be
checked.
Can Change Night Default = Off.
Service Group
If selected, the user can change the fallback group used when the hunt group is in Night
Service mode.
Can Change Out Default = Off. If selected, the user can change the fallback group used when the hunt
of Service Group group is in Out of Service mode.

Related links
Menu Programming on page 466

4400/6400
Navigation: User | Menu Programming | 4400/6400
4412, 4424, 4612, 4624, 6408, 6416 and 6424 phones have a Menu key, sometimes marked with
an icon. When Menu is pressed, a number of default functions are displayed. The < and >
keys can be used to scroll through the functions while the keys below the display can be used to
select the required function.
The default functions can be overwritten by selections made within this tab.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Menu No. The menu position which the function is being set.
Label This is a text label for display on the phone. If no label is entered, the default label for the
selected action is used. Labels can also be changed through the menu on some phones,
refer to the appropriate telephone user guide.
Action Defines the action taken by the menu button.
Table continues…

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User

Field Description
Action Data This is a parameter used by the selected action. The options here will vary according to
the selected button action.

Related links
Menu Programming on page 466

Mobility
Navigation: User | Mobility
The symbol indicates that the setting can also be set by the user rights with which the user
is associated. The user rights can be controlled by a time profile selected as the user's Working
Hours Time Profile setting.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Configuration settings
Twinning allows the IP Office to present a user's calls to both their main phone and another
extension or number. The IP Office system supports two modes of twinning:
Internal Mobile
Twinning Destination Internal extensions on the same IP External numbers only.
Office.
Supported in All locales. All locales.
License Required The primary phone user must be a Yes
licensed user.

Using both internal and mobile twinning

For IP Office R11.1.3 and higher, you can configure both Internal Twinning and Mobile Twinning
for Avaya Workplace Client users:
• The Avaya Workplace Client users can switch between internal and mobile twinning using the
client's Incoming Call Features menu.
• When the Avaya Workplace Client user selects mobile twinning, the internal twinning
extension temporarily reverts to its original extension number. Therefore, Avaya recommend
restricting the internal twin extension to internal calls when not twinned.
Simultaneous
These settings apply to the operation of simultaneous clients.

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 Mobility

Field Description
Coverage Delay Default = 0 seconds. Range= minimum 0 seconds to maximum 15 seconds.
(secs)
Sets the delay between calls alerting on the user's main telephony device/client and then
also alerting their MS Teams client.
MS Teams URI The user's telephony URI for MS Teams. The maximum length of the URI is 161
characters. For more details, see the Deploying MS Teams Direct Routing with IP Office
manual.
This field is read-only if the Auto Populate MS Teams Data setting (System >
Telephony > MS Teams) is enabled.

Internal Twinning
Select this option to enable internal twinning for a user. Internal twinning is not supported during
resilience.
Field Description
Twinned Handset Default = Blank.
This drop-down list is used to select twinned phone. Supported internal twining
destinations must:
• Be on the same IP Office system
• Not be using simultaneous mode.
• Be a physical deskphone or DECT extension. Softphones are not supported.
If the list is grayed out, the user is a twinning destination and the main phone to which it is
twinned is displayed.
All User | Mobility fields are grayed out for unlicensed users.
Maximum Number Default = 1.
of Calls
Sets the number of calls the user can have internally twinned at the same time:
• If set to one, when either the main or twinned phone are in use, any additional incoming
call receives busy treatment.
• If set to two, when either phone is in use, it receives call waiting indication for the
second call. Any further calls above two receive busy treatment.
Twin Bridge Default = Off.
Appearances
Set whether calls alerting on bridged appearance buttons on the main phone also alert on
the twinned phone.
Twin Coverage Default = Off.
Appearances
Set whether calls alerting on coverage appearance buttons on the main phone also alert
on the twinned phone.
Twin Line Default = Off.
Appearances
Set whether calls alerting on line appearance buttons on the main phone also alert on the
twinned phone.

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User

Mobility Features
If enabled, this option allows any of the mobility features to be enabled for the user.
Field Description
Mobile Twinning If selected, the user is enable for mobile twinning. The user can control this option
through a Twinning programmable button on their a phone.
Fallback Twinning Default = Disabled
When enabled , if the user’s main extension is unreachable, the IP Office redirects their
calls to the Twinned Mobile Number even if Mobile Twinning is disabled. Fallback
Twinning does not use the Mobile Dial Delay.
Twinned Mobile Default = Blank.
Number
This field sets the external destination number for mobile twinned calls. The number
is subject to short code processing and should include any external dialing prefix if
necessary.
Twinning Time Default = <None> (Any time)
Profile
This field allows selection of a time profile during which mobile twinning is used.
Mobile Dial Delay Default = 2 seconds
This setting controls how long calls alert at the user's main extension before also alerting
at the twinned number. You can use this setting at the user's request, however you may
also need to use it in some scenarios. For example:
• If the twinning number is a switched off mobile device, the mobile service provider
may immediately answer the call using their voicemail service. That creates a scenario
where the user's main extension does not ring or rings briefly.
Mobile Answer Default = 0 (Off). Range = 0 to 99 seconds.
Guard
This control can be used in situations where calls sent to the twinned destination are
automatically answered by a voicemail service or automatic message if the twinned
device is not available. If a twinned call is answered before the Mobile Answer Guard
expires, the system will drop the call to the twin.
Hunt group calls Default = Off
eligible for mobile
twinning This setting controls whether hunt group calls ringing the user's primary extension should
also be presented to the mobile twinning number.
Forwarded calls Default = Off This setting controls whether calls forwarded to the user's primary
eligible for mobile extension should also be presented to the mobile twinning number.
twinning
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 Mobility

Field Description
Twin When Default = Off.
Logged Out
If enabled, if the user logs off their main extension, calls to that extension will still alert at
their twinned number rather than immediately going to voicemail or busy.
When logged out but twinned:
• Mobile Dial Delay is not applied.
• Hunt group calls (all types) are twinned if Hunt group calls eligible for mobile
twinning is enabled. The user's idle time is reset for each externally twinned call
answered. Note the IP Office automatically treats calls twinned over analog and analog
emulation trunks as answered.
• When the user's Mobile Time Profile is not active, calls are treated the same as the
user was logged out user with no twinning.
• Callback calls initiated by the user will ring the twinned number. Other users can set
automatic callback to the user. The twinned user's busy/free state is tracked for all calls
through the IP Office system.
• The user's bridged appearance buttons do not alert. Their coverage appearance
buttons will continue to operate.
• The BLF/user button status shown for the user is:
- For calls alerting or in progress through the IP Office system to the twin, the user
status shows alerting or in-use. The user shows as busy/in-use if they such a call on
hold and they have Busy on Held enabled.
- If the user enables DND through Mobile Call Control, their status shows as DND/busy.
- Calls from the IP Office system dialed direct to the user's twinned destination rather
than redirected by twinning do not change the user's status.
one-X Mobile Default = Off.
Client
Not supported with R11.1 and higher.
Mobile Call Default = Off.
Control
This feature allows a user receiving a call on their twinned device to access system dial
tone and then perform dialing action including making calls and activating short codes.
See Mobile Call Control on page 776.
Mobile Callback Default = Off.
Mobile callback allows the user to make calls from the twinned number using the IP Office
to route the calls. See Mobile Call Control on page 776.
When used:
• The user calls the IP Office system and then hangs up.
• The IP Office system calls the user's caller ID number.
• When answered, the IP Office provide dial tone for the user to make a call.

Related links
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User

Group Memberships
Navigation: User | Group Membership
This tab displays the hunt group of which the user has been made a member. The tick boxes
indicate whether the user's membership of each of those groups is currently enabled or disabled.
Related links
User on page 433

Announcements
Navigation: User | Announcements
Announcements are played to callers waiting to be answered. This includes callers being
presented to hunt group members, ie. ringing, and callers queued for presentation.
• The system supports announcements using Voicemail Pro or Embedded Voicemail.
• If no voicemail channel is available for an announcement, the announcement is not played.
• In conjunction with Voicemail Pro, the system allows a number of voicemail channels to be
reserved for announcements. See System | Voicemail.
• With Voicemail Pro, the announcement can be replaced by the action specified in a Queued
(1st announcement) or Still Queued (2nd announcement) start point call flow. Refer to the
Voicemail Pro Installation and Maintenance documentation for details.
• Calls can be answered during the announcement. If it is a mandatory requirement that
announcements should be heard before a call is answered, then a Voicemail Pro call flow
should be used before the call is presented.

Note:
Call Billing and Logging
A call becomes connected when the first announcement is played to it. That connected
state is signaled to the call provider who may start billing at that point. The call will also
be recorded as answered within the SMDR output once the first announcement is played.
• If a call is rerouted, for example forwarded, the announcement plan of the original user is still
applied until the call is answered. The exception is calls rerouted to a hunt group at which
point the hunt group announcement settings are applied.
• For announcements to be used effectively, either the user's no answer time must be
extended beyond the default 15 seconds or Voicemail On should be deselected.
Recording Announcements
Voicemail Pro:
There is no mechanism within the telephony user interfaces (TUI) to record user announcements.
To provide custom announcements, user queued and still queued start points must be configured
with Voicemail Pro with the required prompts played by a generic action.

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 Announcements

Embedded Voicemail:
Embedded Voicemail does not include any default announcement or method for recording an
announcement. The Record Message short code feature is provided to allow the recording
of announcements. The telephone number field of short codes using this feature requires the
extension number followed by either ".1" for announcement 1 or ".2" for announcement 2. For
example, for extension number 300, the short codes *91N# | Record Message | N".1" and *92N#
| Record Message | N".2" could be used to allow recording of the announcements by dialing
*91300# and *92300#.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Announcements Default = Off.
On
This setting enables or disables announcements.
Wait before 1st Default = 10 seconds. Range = 0 to 255 seconds.
announcement:
This setting sets the time delay from the calls presentation, after which the first
announcement should be played to the caller.
Flag call as Default = Off.
answered
This setting is used by the CCC and CBC applications. By default they do not regarded a
call as answered until it has been answered by a person or by a Voicemail Pro action with
Flag call as answered selected. This setting allows calls to be marked as answered
once the caller has heard the first announcement.
Post Default = Music on hold.
announcement
Following the first announcement, you can select whether the caller should hear Music on
tone
Hold, Ringing or Silence until answered or played another announcement.
2nd Default = On.
Announcement
If selected, a second announcement can be played to the caller if they have still not been
answered.
Wait before 2nd Default = 20 seconds. Range = 0 to 255 seconds.
announcement
This setting sets the wait between the 1st and the 2nd announcement.
Repeat last Default = On.
announcement
If selected, the last announcement played to the caller is repeated until they are
answered or hang-up.
Wait before repeat Default = 20 seconds. Range = 0 to 255 seconds.
If Repeat last announcement is selected, this setting sets is applied between each
repeat of the last announcement.

Related links
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User

SIP
Navigation: User | SIP
This tab is available when either of the following has been added to the configuration:
• an IP Office Line
• a SIP trunk with a SIP URI record containing a field that has been set to Use Internal Data.
Various fields within the URI settings used by SIP trunks can be set to Use Internal Data. When
that is the case, the values from this tab are used into the URI when the user makes or receives
SIP calls. Within a multi-site network, that includes calls which break out using a SIP trunk on
another system within the network.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
SIP Name Default = Blank on Voicemail tab/Extension number on other tabs.
This value is used for fields, other the Contact header, where the SIP URI entry being
used has its Contact field set to Use Internal Data.
• On incoming calls, if the Local URI is set to Use Internal Data, the system can
potentially match the received R-URI or From header value to a user and/or group SIP
Name. This requires the SIP URIs Incoming Group to match a Incoming Call Route
with the same Line Group ID and a . (period) destination.
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display field of the SIP URI being used is set to
Use Internal Data.
Contact Default = Blank on Voicemail tab/Extension number on other tabs.
The value is used for the Contact header when the Contact field of the SIP URI being
used for a SIP call is set to Use Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs.
If the From field in the SIP URI is set to Use Internal Data, selecting this option inserts
Anonymous into that field rather than the SIP Name set above. See Anonymous SIP
Calls on page 842.

Related links
User on page 433

Personal Directory
Navigation: User | Personal Directory

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 Personal Directory

Each user is able to have up to 250 personal directory records up to the overall system limit.
Those records are used as follows:
• When using M-Series, T-Series, 1400, 1600, 9500, 9600 or J100 Series phones, the user is
able to view and call their personal directory numbers.
• When using a 1400, 1600, 9500, 9600 or J100 Series phone, the user is also able to edit and
add personal directory records.
• On phones that support hot desking on the same system or to another system in a multi-site
network, the user can still access their personal directory.
Users are able to view and edit their personal directory through their phone. Directory records are
used for dialing and caller name matching.
Directory Dialing
Directory numbers are displayed by user applications such as SoftConsole. Directory numbers are
viewable through the Dir function on many Avaya phones (Contacts or History). They allow the
user to select the number to dial by name. The directory will also contain the names and numbers
of users and hunt groups on the system.
The Dir function groups directory records shown to the phone user into the following categories.
Depending on the phone, the user may be able to select the category currently displayed. In
some scenarios, the categories displayed may be limited to those supported for the function being
performed by the user:
• External - Directory records from the system configuration. This includes HTTP and LDAP
imported records.
• Groups - Groups on the system. If the system is in a multi-site network, it will also include
groups on other systems in the network.
• Users or Index - Users on the system. If the system is in a multi-site network it will also
include users on other systems in the network.
• Personal - Available on 1400, 1600, 9500, 9600 and J100 Series phones. These are the
user's personal directory records stored within the system configuration.
Speed Dialing
On M-Series and T-Series phones, a Speed Dial button or dialing Feature 0 can be used to
access personal directory records with an index number.
• Personal: Dial Feature 0 followed by * and the 2-digit index number in the range 01 to 99.
• System: Dial Feature 0 followed by 3-digit index number in the range 001 to 999.
• The Speed Dial short code feature can also be used to access a directory speed dial using its
index number from any type of phone.
Caller Name Matching
Directory records are also used to associate a name with the dialed number on outgoing calls or
the received CLI on incoming calls. When name matching is being done, a match in the user's
personal directory overrides any match in the system directory. Note that some user applications
also have their own user directory.
SoftConsole applications have their own user directories which are also used by the applications
name matching. Matches in the application directory may lead to the application displaying a
different name from that shown on the phone.

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User

Name matching is not performed when a name is supplied with the incoming call, for example
QSIG trunks. On SIP trunks the use of the name matching or the name supplied by the trunk can
be selected using the setting System | Telephony | Telephony | Default Name Priority. This
setting can also be adjusted on individual SIP lines to override the system setting.
Directory name matching is not supported for DECT handsets. For information on directory
integration, see IP Office DECT R4 Installation.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Index Range = 00 to 99 or None.
This value is used with personal speed dials set and dialed from M and T-Series phones.
The value can be changed but each value can only be applied to one directory record
at any time. Setting the value to None makes the speed dial inaccessible from M and
T-Series phones, however it may still be accessible from the directory functions of other
phones and applications. The Speed Dial short code feature can be used to create short
codes to dial the number stored with a specific index value. Release 10.0 allows users to
have up to 250 personal directory entries. However, only 100 of those can be assigned
index numbers.
Name Range = Up to 31 characters.
Enter the text to be used to identify the number.
Number Range = Up to 31 digits plus * and #. Enter the number, without spaces, to be dialed.
Wildcards are not supported in user personal directory records. Note that if the system
has been configured to use an external dialing prefix, that prefix should be added to
directory numbers.

Related links
User on page 433

User Portal
Navigation: User | Web Self Administration
Use this menu to enable user portal for a user. You can configure whether they can use user portal
and what features they can access within the user portal menus. For a user guide, refer to the
Using the IP Office User Portal.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Name Description
Enable User Portal Default = Off
When enabled, the user can log into user portal by entering the address of the system
in the format http://<address> and then selecting IP Office User Portal. The login
uses the user’s User Name and Password.
Table continues…

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 User Portal

Name Description
Run Enduser Wizard Default = Off
If enabled, the user is walked through a series of menus when they login for the first
time.
Allowed Call Default = Both
Operations
Set whether and how the user can use their user portal to make and answer calls.
The user can change the current mode through their portal's Profile menu. The 'user
choice' column in the table below indicates the options that the user can select and the
default option used when they log in to the portal.
Note that modes other than None are only supported by users with the following
licensed/subscribed profiles:
• On subscription systems, Telephony Plus User and UC User users.
• On non-subscription systems, Power User users.
All systems support the following modes:

Admin Setting Description User Choice

None Do not use the portal to control current None
calls.
Call Control Use the user portal to control calls using None
the user's deskphone.
Call Control[1]

Linux-based IP Office systems also support the following additional modes:

Admin Setting Description User Choice

Softphone[2] Use the user's portal as a WebRTC None
softphone. Call audio uses the browser's
Softphone[1][2]
speaker and microphone settings.
Both Support any of the call operation modes. None
Call Control[1]
Softphone[2]

1. This is the default mode the client will start in.

2. Softphone mode uses WebRTC provided by the IP Office system. For remote
portal users, additional configuration of STUN or TURN is also required. See the
notes at the bottom of the page.

User Settings Access

These options control the options that the user can access within self-administration and the type
of access they have. For each set of options, the user can be given the following access:
• No Access - The user cannot access the related menu and its settings.
• Read Access - The user can view the settings on the menu but cannot change them.
• Write Access - The user can both view and change the settings on the menu.

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User

Name Description
Profile This menu provides the access to details such as full name, voicemail and login code
and email address.
Call Handling This menu provides access to call controls such as forwarding, do not disturb and
twinning.
Personal Directory This menu provides access to the user's personal directory entries.
Button Programming This option allows the user to assign features to programmable buttons on their phone
and to change button labels. They still cannot override the settings of appearance
buttons and buttons set by user rights.
Download This option display a menu of links for user applications that work with IP Office. Note
Applications that the user may require further configuration to use a specific application.

Media Manager Replay Self-Administration

These settings control the users rights to play call recordings stored by Media Manager or
Centralized Media Manager.
Name Description
Enable Media Default = Off.
Manager Replay
When enabled, the user can replay call recordings through web self-administration.
• Note: For users where Media Manager is provided by a separate application server,
recordings are viewed and accessed using the address of the application server
rather than that of the IP Office system.
Replay All If selected, the user can view and replay all recordings.
Recordings
Replay Own If selected, the user can view and replay their own call recordings. When enabled, the
Recordings Replay Recordings For Group and Replay Recordings For Others options are also
available.
Replay Recordings These menus allows the selection of groups for which the user is able to view and
For Group replay recordings.
Replay Recordings The field can be used to enter a list of numbers, separated by semi-colons, for which
For Others the user can view and playback recordings. Those numbers can be accounts codes,
line numbers, user extensions and group extension numbers. The list can be 127
characters in length.
Download If selected, the user is able to download recordings as a separate file.
Recordings
• Downloaded files are outside of the control of the system. Therefore, if you allow
users to download files, it is your responsibility to ensure that they comply with local
privacy and data protection laws regarding the use of those files.

Historical Call Reporting

Call reporting allows the user to view a summary of recent calls by all users. This is currently
a trial feature. It is only supported with subscription mode systems. The system must have its
System > SMDR set to Hosted Only.

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 User Portal

Name Description
Enable Historical Call Default = Off.
Reporting
When enabled, the user can access the call reporting menus through their user portal.
For details, refer to the Using the IP Office Embedded Call Reporter manual.

User Portal Softphone Remote Access Notes

Non-IP500 V2 IP Officesystems can support the user portal as a WebRTC softphone. When
operating as a remote extension, this may require the following:
• The IP Office and user portal to use STUN.
• Connection using an SBC configured for TURN.
For details, see the System > LAN1 > Network Topology > WebRTC settings.
Related links
User on page 433

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Chapter 30: Group

Additional configuration information

This section provides the Group field descriptions. For additional configuration information, see
Group Operation on page 765.
Related links
Group on page 480
Queuing on page 485
Overflow on page 488
Fallback on page 490
Voicemail on page 493
Voice Recording on page 498
Announcements on page 499
SIP on page 502

Group
Navigation: Group | Group
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Configuration settings
The Group settings are used to define the name, extension number and basic operation of the
group. It is also used to select the group members.
You can merge these settings without needing to reboot the IP Office.

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 Group

Field Description
Name Range = Up to 15 characters
The name to identify this group. This field is case sensitive and must be unique.
• Do not start names with a space. Do not use punctuation characters such as #, ?, /, ^,
> and ,.
• Voicemail uses the name to match a group and its mailbox. Changing the name will
route voicemail calls to a new mailbox. Note that Voicemail Pro is not case-sensitive.
For example it will treat "Sales", "sales" and "SALES" as being the same.
Profile Default = Standard Hunt Group
Defines the group type. The options are:

Profile Description
Standard Hunt The default group type and the standard method for creating IP
Group Office user groups.
XMPP Group Extensible Messaging and Presence Protocol (XMPP) is a
communications protocol for presence status and Instant Messaging
(IM). Select XMPP to enable presence information and instant
messaging within a defined group of XMPP enabled one-X clients.
Two users can see each other's presence and exchange instant
messages only if they are members of the same XMPP group. A
user can be a member of zero or more groups.

Important:
Before adding a user to an XMPP group, the user must be
added to the configuration and the configuration saved. If the
user is added to the group before the directory is synchronized,
the user will not be visible in one-X Portal.
Centralized Used for centralized extensions that are normally handled by the
Group core feature server (Avaya Aura®) and are handled by the IP Office
only when in survival mode due to loss of connection to the Avaya
Aura®.
Calls arriving to a centralized hunt group number when the Avaya
Aura® line is in-service are sent by the IP Office to Avaya Aura®.
Calls arriving to a centralized hunt group number when the Avaya
Aura Session Manager line is out-of-service are processed by the IP
Office and targeted to the hunt group members as configured on the
IP Office.
Extension Range = 1 to 15 digits.
This sets the directory number for calls to the hunt group.
• Groups for CBC and CCC should only use up to 4 digit extension numbers.
• Extension numbers in the range 8897 to 9999 are reserved for use by the IP Office
Delta Server.
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Group

Field Description
Exclude From Default = Off
Directory
When on, the user does not appear in the directory list shown by the user applications
and on phones with a directory function.
Ring Mode Default = Sequential
Sets how the system determines which hunt group member to ring first and the next hunt
group member to ring if unanswered. This is used in conjunction with the User List which
list the order of group membership. The options are:

Ring Mode Description

Collective All available phones in the User List phones ring simultaneously.
Collective Call This is a Collective hunt group as above but with hunt group call
Waiting waiting also enabled. When an additional call to the hunt group call is
waiting to be answered, users in the group who are already on a call
will receive call waiting indication.
• On phones with call appearance buttons, the call waiting indication
takes the form of an alert on the next available call appearance
button. The user's own Call Waiting On setting is overridden when
they are using a phone with call appearances.
• On other phones, call waiting indication is given by a tone in the
speech path (the tone is locale specific). The user's Call Waiting
On setting is used in conjunction with the hunt group setting.
Sequential Each extension is rung in order, one after the other, starting from the
first extension in the list each time.
Rotary Each extension is rung in order, one after the other. However, the last
extension used is remembered. The next call received rings the next
extension in the list.
Longest The extension that has been unused for the longest period rings first,
Waiting then the extension that has been idle second longest rings, etc. For
extensions with equal idle time, 'sequential' mode is used.
Where hunt group calls are being presented to a twinned extension,
the longest waiting status of the user can be reset by calls answered
at either their master or twinned extension.
No Answer Time Default = System Default. Range = System Default or 6 to 99999 seconds.
(secs)
The number of seconds an extension rings before the call is passed to another extension
in the list. This applies to all telephones in this group and also any Overflow Group List
groups it uses. Leave blank to use the system default setting (System > Telephony >
Telephony > Default No Answer Time).
• This does not apply for collective hunt groups, where calls will continue ringing until
either the Overflow Time or Group No Answer Time applies.
• If the group contains users who are using Avaya Workplace Client on iOS devices,
Avaya recommends the time is set to at least 20 seconds.
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 Group

Field Description
Hold Music Source Default = No Change.
The system can support multiple music on hold sources; the System Source (either an
internal file or the external source port or tones) plus a number of additional internal
sources (3 on IP500 V2 systems, 31 on Linux systems), see System > Telephony >
Tones & Music.
Before reaching a hunt group, the source used is set by the system wide setting or by the
Incoming Call Route that routed the call. If the system has several hold music sources
available, this field allows selection of the source to associate with calls presented to this
hunt group or to leave it unchanged. The new source selection will then apply even if the
call is forwarded or transferred out of the hunt group unless changed again by another
hunt group.
If the call is routed to another system in a multi-site network, the matching source on that
system is used if available.
Hunt group calls overflowing ignore the settings of the Overflow Group List groups.
Calls going to night service or out of service fallback group use the hold music source
setting of the original hunt group and then, if different, the setting of the fallback group.
The setting of further fallback groups from the first are ignored.
Ring Tone Override Default = Blank
If ring tones have been configured in the System | Telephony | Ring Tones tab, they
are available in this list. Setting a ring tone override applies a unique ring tone for the
hunt group. Ring tone override features are only supported on 1400 Series, 9500 Series
and J100 Series (except J129) phones.
Agent's Status on Default = None (No status change).
No-Answer Applies
For hunt group members with a login code set and Force Log enabled, the system can
To
change their status if they do not answer a hunt group call presented to them within the
group's No Answer Time.
• This setting defines what type of hunt group calls can trigger use of the agent's Status
on No Answer setting. The options are None, Any Call and External Inbound Calls
Only.
• The new status is set by the agent's Status on No Answer setting (User >
Telephony > Supervisor Settings).
• The Status on No Answer action does not apply if the call is presented and then
answered elsewhere or the caller disconnects.
• This option is not used for calls ringing the agent because the group is in another
group's Overflow Group List.
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Group

Field Description
User List This is an ordered list of the users who are members of the hunt group. For Sequential
and Rotary groups it also sets the order in which group members are used for call
presentation.
• Repeated numbers can be used, for example 201, 202, 201, 203, etc. Each extension
will ring for the number of seconds defined by the No Answer Time before moving to
the next extension in the list, dependent on the Hunt Type chosen.
• The check box next to each member indicates the status of their membership. Group
calls are not presented to members who have their membership currently disabled.
However, those users are still able to perform group functions such as group call
pickup.
• The order of the users can be changed by dragging the existing records to the required
position.
• To add records select Edit. A new menu is displayed that shows available users on the
left and current group members of the right. The lists can be sorted and filtered.
• Users on remote systems in a multi-site network can also be included. Groups
containing remote members are automatically advertised within the network.
• Before adding a user to an XMPP group, the user must be added to the configuration
and the configuration saved. If the user is added to the group before the directory is
synchronized, the user will not be visible in one-X Portal.

Related links
Group on page 480
User List Select Members on page 484

User List Select Members

Navigation: Group | Group | Edit
To open the Select Memebers window, click Edit.
The hunt group Select Members form is used to add and remove users from the hunt group. For
hunt group's with a Ring Mode of Sequential or Rotary it is also used to set the order of use for
the members of the hunt group.
The filters section at the top of the form can be used to filter the users shown. Note for hunt
groups set as a CCR Agent Group, only users set as CCR Agent are shown.
The controls and data on the form vary depending on the hunt group's Ring Mode setting and on
whether the system is in a multi-site network.
To sort either table, click on the column header that should be used for the sort the table. Sort
changes the order of display only, it does not change the actual order of hunt group membership.
For Sequential and Rotary hunt groups, an Order column is shown. To change the order position
of a hunt group member, select the member and then use the up and down arrow buttons.

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 Queuing

During the actions below, the Shift and Ctrl keys can be used as normal to select multiple
users. Note that the list of members has been sorted, the sort is updated after adding or moving
members.
• Add Before Using the Shift and/or Ctrl keys, select the users you want to add and then on
the right select the existing member that you want to add them before.
• Add After Using the Shift and/or Ctrl keys, select the users you want to add and then on the
left select the existing member after which you want them added.
• Append Add the selected users on the left to the hunt group members on the right as the last
member in the group order.
• Remove Remove the selected users on the right from the list of hunt group members.
• Move the selected member on the right up or down the membership order of the group.
Related links
Group on page 480

Queuing
Navigation: Group | Queuing
Any calls waiting to be answered at a hunt group are regarded as being queued. The Normalise
Queue Length control allows selection of whether features that are triggered by the queue length
should include or exclude ringing calls. Once one call is queued, any further calls are also queued.
When an available hunt group member becomes idle, the first call in the queue is presented. Calls
are added to the queue until the hunt group's Queue Limit, if set, is reached.
• When the queue limit is reached, any further calls are redirected to the hunt group's
voicemail if available.
• If voicemail is not available excess calls receive busy tone. An exception to this are analog
trunk and T1 CAS trunk calls which will remain queued regardless of the queue limit if no
alternate destination is available.
• If an existing queued call is displaced by a higher priority call, the displaced call will remain
queued even if it now exceeds the queue limit.
Hunt group announcements are separate from queuing. Announcements can be used even
if queuing is turned off and are applied to ringing and queued calls. See Hunt Group |
Announcements.
There are several methods of displaying a hunt group queue.
• Group Button: On phones, with programmable buttons, the Group function can be assigned
to monitor a specified group. The button indicates when there are calls ringing within the
group and also when there are calls queued. The button can be used to answer the longest
waiting call.
• SoftConsole: The SoftConsole applications can display queue monitors for up to 7 selected
hunt groups. This requires the hunt group to have queuing enabled. These queues can be
used by the SoftConsole user to answer calls.

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Group

When a hunt group member becomes available, the first call in the queue is presented to that
member. If several members become available, the first call in the queue is simultaneously
presented to all the free members.
Overflow Calls Calls that overflow are counted in the queue of the original hunt group from which
they overflow and not that of the hunt group to which they overflow. This affects the Queue Limit
and Calls in Queue Threshold.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Queuing On Default = On
If enabled, calls to the hunt group are queued.
Queue Length Default = No Limit. Range = No Limit, 1 to 99 calls.
This setting can be used to limit the number of calls that can be queued. Calls exceeding
this limit are passed to voicemail if available or otherwise receive busy tone. This value is
affected by Normalize Queue Length setting.
• If voicemail is not available excess calls receive busy tone. An exception to this is
analog trunk and T1 CAS trunk calls which will remain queued regardless of the queue
limit if no alternate destination is available. This is due to the limited call status signalling
supported by those trunks which would otherwise create scenarios where the caller
has received ringing from the local line provider and then suddenly gets busy from the
system, creating the impression that the call was answered and then hung up.
• If priority is being used with incoming call routes, high priority calls are place ahead of
lower priority calls. If this would exceed the queue limit the limit is temporarily increased
by 1.
• If an existing queued call is displaced by a higher priority call, the displaced call will
remain queued even if it now exceeds the queue limit.
Normalize Queue Default = On.
Length
Calls both waiting to ring and ringing are regarded as being queued. This therefore affects
the use of the Queue Limit and Calls in Queue Alarm thresholds. If Normalize Queue
Length is enabled, the number of hunt group members logged in and not on DND is
added to those thresholds.
For example, a customer has two products that it is selling through a call center with 10
available agents; one product with a $10 margin and one with a $100 margin. Separate
hunt groups with the same 10 members are created for each product.
• The $100 product has a Queue Limit of 5 and Normalize Queue Length is on. The
maximum number of $100 calls that can be waiting to be answered will be 15 (10
ringing/connected + 5 waiting to ring).
• The $10 product has a Queue Limit of 5 and Normalize Queue Length is off. The
maximum number of $10 calls that can be waiting to be answered is 5 (5 ringing/
connected).
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 Queuing

Field Description
Queue Type Default = Assign Call On Agent Answer.
When queuing is being used, the call that the agent receives when they answer can be
assigned in one of two ways:
• Assign Call On Agent Answer In this mode the call answered by the hunt group
member will always be the longest waiting call of the highest priority. The same call will
be shown on all ringing phones in the group. At the moment of answering that may not
necessarily be the same call as was shown by the call details at the start of ringing.
• Assign Call on Agent Alert In this mode, once a call has been presented to a hunt
group member, that is the call they will answer if they go off hook. This mode should be
used when calls are being presented to applications which use the call details such as a
fax server, CTI or TAPI.
Calls In Queue The system can be set to send an alert to a analog specified extension when the number
Alarm of calls queued for the hunt group reaches the specified threshold.
Calls In Queue Default = Off. Range = 1 to 99.
Threshold
Alerting is triggered when the number of queued calls reaches this threshold. Alerting will
stop only when the number of queued calls drops back below this threshold. This value is
affected by Normalize Queue Length setting above.
Analog Extension Default = <None>.
to Notify
This should be set to the extension number of a user associated with an analog extension.
The intention is that this analog extension port should be connected to a loud ringer or
other alerting device and so is not used for making or receiving calls. The list will only
shown analog extensions that are not members of any hunt group or the queuing alarm
target for any other hunt group queue. The alert does not follow user settings such as
forwarding, follow me, DND, call coverage, etc or receive ICLID information.

Group Queue Controls

Group Queue Settings
Manager Hunt group queuing is enabled using the Queuing On option on the Hunt Group |
Queuing tab.
Controls The following short code features/button programming actions can be used:

SoftConsole SoftConsole can display up to 7 hunt group queues (an eighth queue is reserved for recall
calls). They are configured by clicking and selecting the Queue Mode tab.
• Within the displayed queues, the number of queued calls is indicated and the time of the
longest queued call is shown. Exceeding an alarm threshold is indicated by the queue
icons changing from white to red. The longest waiting call in a queue can be answered
by clicking on the adjacent button.
• For each queue, an alarm threshold can be set based on number of queued calls and
longest queued call time. Actions can then be selected for when a queue exceeds its
alarm threshold; Automatically Restore SoftConsole, Ask me whether to restore
SoftConsole or Ignore the Alarm.

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Group

Related links
Group on page 480

Overflow
Navigation: Group | Overflow
Overflow can be used to expand the list of group members who can be used to answer a call. This
is done by defining an overflow group or groups. The call is still targeted to the original group and
subject to that group's settings, but is now presented to available members in the overflow groups
in addition to its own available members.
Overflow calls still use the settings of the original target group. The only settings of the overflow
group that is used is it's Ring Mode. For example:
• Calls that overflow use the announcement settings of the group from which they are
overflowing.
• Calls that overflow use the Voicemail Answer Time of the original group from which are are
overflowing.
• Calls that are overflowing are included in the overflowing group's Queue Length and Calls
In Queue Threshold. They are not included in those values for the hunt group to which they
overflow.
• The queuing and overflow settings of the overflow groups are not used, ie. calls cannot
cascade through a series of multiple overflows.
A call will overflow in the following scenarios:
• If Queuing is off and all members of the hunt group are busy, a call presented to the group
will overflow immediately, irrespective of the Overflow Time.
• If Queuing is on and all members of the hunt group are busy, a call presented to the group
will queue for up to the Overflow Time before overflowing.
• If Queuing is on but there are no members logged in or enabled, calls can be set to overflow
immediately by setting the Overflow Immediate setting to No Active Members. Otherwise
calls will queue until the Overflow Time expires.
• If no Overflow Time is set, a call will overflow when it has rung each available hunt group
member without being answered.
• Once one call is in overflow mode, any additional calls will also overflow if the Overflow
Mode is set to Group (the default).
An overflow call is presented to available group members as follows:
• Once a call overflows, it is presented to the first available member of the first overflow group
listed. The Ring Mode of the overflow group is used to determine its first available member.
However the No Answer Time of the original targeted group is used to determine how long
the call is presented.
• When the No Answer Time expires, the call is presented to the next available member in
the overflow group. If all available members in the overflow group have been tried, the first
member in the next listed overflow group is tried.

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 Overflow

• When the call has been presented to all available members in the overflow groups, it is
presented back to the first available member in the original target group.
• While the call is being presented to members in an overflow group, the announcement and
voicemail settings of the original targeted group are still applied.
For calls being tracked by the Customer Call Reporter application, overflow calls are recorded
against the original targeted group but using separate statistics; Overflowed Calls, Overflowed
Calls Waiting, Overflowed Answered and Overflowed Lost.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Overflow Time Default = Blank. Range = Off or 1 to 3600 seconds.
For a group using queuing, the Overflow Time sets how long a call queues before being
presented to available agents in the group's Overflow Group List. Note that if the call is
currently ringing an agent when the timer expires, it will complete ringing for the group's
No Answer Time before overflowing.
Overflow Mode Default = Group.
This option allows selection of whether the overflow of queued calls is determined on
an individual call by call basis or is applied to all calls once any one call overflows. The
options are:
• Group: In this mode, once one call overflows all additional queued calls also overflow.
• Call: In this mode, each individual call will follow the group's overflow settings before it
overflows.
Immediate Default = Off.
Overflow:
For groups which are using queueing, this setting can be used to control whether calls
should overflow immediately when there are no available or active agents. The options
are:
• Off: Do not overflow immediately. Use the Overflow Time setting as normal.
• No Active Agents: Overflow immediately if there are no available or active agents as
defined above, regardless of the Overflow Time setting.
- An active agent is an agent who is either busy on a call or in after call work. An
available agent is one who is logged in and enabled in the hunt group but is otherwise
idle.
- A hunt group is automatically treated as having no available or active agents if:
- The group's extension list is empty.
- The group's extension list contains no enabled users.
- The group's extension list contains no extensions that resolve to a logged in agent (or
mobile twin in the case of a user logged out mobile twinning).
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Group

Field Description
Overflow Group This list is used to set the group or groups that are used for overflow. Each group is
List used in turn, in order from the top of the list. The call is presented to each overflow
group member once, using the Ring Mode of the overflow group. If the call remains
unanswered, the next overflow group in the list is used. If the call remains unanswered at
the end of the list of overflow groups, it is presented to available members of the original
targeted group again and then to those in its overflow list in a repeating loop. A group can
be included in the overflow list more than once if required and the same agent can be in
multiple groups.

Related links
Group on page 480

Fallback
Navigation: Group | Fallback
Fallback settings can be used to make a hunt group unavailable and to set where the hunt group's
calls should be redirected at such times. Hunt groups can be manually placed In Service, Out of
Service or in Night Service. Additionally using a time profile, a group can be automatically placed
in Night Service when outside the Time Profile settings.
Fallback redirects a hunt group's calls when the hunt group is not available, for example outside
normal working hours. It can be triggered either manually or using an associated time profile.
Group Service States:
A hunt group can be in one of three states; In Service, Out of Service or Night Service. When In
Service, calls are presented as normal. In any other state, calls are redirected as below.
Call Redirection:
The following options are possible when a hunt group is either Out of Service or in Night
Service.
• Destination: When in Out of Service, if an Out of Service Destination has been set, calls
are redirected to that destination. When in Night Service, if a Night Service Destination
has been set, calls are redirected to that destination.
• Voicemail: If no fallback destination has been set but voicemail is enabled for the group,
calls are redirected to voicemail.
• Busy Tone: If no fallback destination has been set and voicemail is not available, busy tone
is returned to calls.
Manually Controlling the Service State:
Manager and or short codes can be used to change the service state of a hunt group. The short
code actions can also be assigned to programmable buttons on phones.

• The icon is used for a hunt group manually set to Night Service mode.

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 Fallback

• The icon is used for a hunt group manually set to Out of Service mode.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported. You can manually override a time profile.
Time Profile:
A Day Service Time Profile can be associated with the hunt group. A time profile if required, is
set through Time Profile | Time Profile.
When outside the time profile, the hunt group is automatically placed into night service. When
inside the time profile, the hunt group uses manually selected mode.
• When outside the time profile and therefore in night service, manual night service controls
cannot be used to override the night service. However the hunt group can be put into out of
service.
• When a hunt group is in Night Service due to a time profile, this is not indicated within
Manager.
• Time profile operation does not affect hunt groups set to Out of Service.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Day Service Time Default = <None> (No automatic night service)
Profile
This field allows selection of a previously created Time Profile. That profile then specifies
the times at which it should use the manually selected Service Mode settings. Outside the
period defined in the time profile, the hunt group behaves as if set to Night Service mode.
Note that when a hunt group is in Night Service due to it associated time profile, this is not
reflected by the Service Mode on this tab. Note also that the manual controls for changing
a hunt group's service mode cannot be used to take a hunt group out of time profile night
service.
Night Service Default = <None> (Voicemail or Busy Tone)
Destination
This field sets the alternate destination for calls when this hunt group is in Night Service
mode. The destination can be a group, a user, a short code, or an Auto Attendant. Select
a group or user from the drop down list. Manually enter a short code or an Auto Attendant
name.
If left blank, calls are redirected to voicemail if available or otherwise receive busy tone.
Out of Service Default = <None> (Voicemail or Busy Tone)
Fallback Group
This field sets the alternate destination for calls when this hunt group is in Out of Service
mode. The destination can be a group, a user, a short code, or an Auto Attendant. Select
a group or user from the drop down list. Manually enter a short code or an Auto Attendant
name. For Auto Attendant names, use the format AA:Name.
If left blank, calls are redirected to voicemail if available or otherwise receive busy tone.
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Group

Field Description
Mode Default = In Service
This field is used to select the current service mode for the hunt group manually. The
options are:
• In Service: When selected, the hunt group is enabled. This is the default mode.
• Night Service: When selected, calls are redirected using the Night Service Fallback
Group setting. This setting can also be manually controlled using the short code, and
button programming features Set Hunt Group Night Service and Clear Hunt Group Night
Service.
• Out of Service: When selected, calls are redirected using the Out of Service Fallback
Group setting. This setting can also be manually controlled using the short code, and
button programming features Set Hunt Group Out of Service and Clear Hunt Group Out
of Service.
Group No Answer Default = 45 seconds, Range = 1 to 3600 seconds.
Time
This setting sets the time duration on presting a call to a hunt group and its overflow
groups if set before going to the group's Group No Answer Destination.
Exceeding the time duration redirects the call regardless of any announcements, overflow,
or queue. If Group No Answer Time is set to Off, the no answer destination is used, and
once each available member of the hunt group is alerted for the group's No Answer Time.
Group No Answer When an unanswered call to a hunt group reaches the Group No Answer Time, you can
Destination configure the following options:
• <NONE> - The destination is not used. Instead, calls continue ringing against the hunt
group.
• Voicemail - The call is redirected to a voicemail to leave a message and uses the call
original destination mailbox. Set to Voicemail for default configurations.
• The drop-down list includes all other group and user extensions and redirects the call to
that extension.
• You can enter a number manually to match against system short codes.

Hunt Group Fallback Controls

The following short code features and button programming actions can be used.
Feature/Action Short Code Default Button
Set Hunt Group Night Yes *20*N# Yes-Toggles
Service
Clear Hunt Group Night Yes *21*N# Yes
Service
Set Hunt Group Out of No No Yes-Toggles
Service
Clear Hunt Group Out of No No Yes
Service

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 Voicemail

Note that for a hunt group using a time profile, these controls only are only applied when the hunt
group is within the specified time profile period. When outside its time profile, the hunt group is in
night service mode and cannot be overridden.
Related links
Group on page 480

Voicemail
Navigation: Group | Voicemail
The system supports voicemail for hunt groups in addition to individual user voicemail mailboxes.
If voicemail is available and enabled for a hunt group, it is used in the following scenarios.
Scenario Description
Group No Answer For 11.1 FP1 and higher, the use of voicemail to answer calls during normal operation is
Time controlled by the group's Fallback settings.
Voicemail Answer This option is only used for pre-11.1 FP1 systems. A call goes to voicemail when this
Time timeout is reached, regardless of any announcement, overflow, queuing or other settings.
The default timeout is 45 seconds.
Unanswered Calls A call goes to voicemail when it has been presented to all the available hunt group
members without being answered. If overflow is being used, this includes be presented
to all the available overflow group members.
Night Service A call goes to voicemail if the hunt group is in night service with no Night Service
Fallback Group set.
Out of Service A call goes to voicemail if the hunt group is out of service with no Out of Service
Fallback Group set.
Queue Limit If queuing is being used, it overrides use of voicemail prior to expiry of the Voicemail
Reached Answer Time, unless the number of queued callers exceeds the set Queue Limit. By
default there is no set limit.
Automatic Call Incoming calls to a hunt group can be automatically recorded using the settings on the
Recording Hunt Group > Voice Recording tab.

When a caller is directed to voicemail to leave a message, the system indicates the target user or
hunt group mailbox.
The mailbox of the originally targeted user or hunt group is used. This applies even if the call
has been forwarded to another destination. It also includes scenarios where a hunt group call
overflows or is in fallback to another group.
Voicemail Pro can be used to customize which mailbox is used separately from the mailbox
indicated by the system.
By default no user is configured to receive message waiting indication when a hunt group
voicemail mailbox contains new messages. Message waiting indication is configured by adding
a H groupname record to a user's SourceNumbers tab (User > Source Numbers).

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Group

By default, no mechanism is provided for access to specific hunt group mailboxes. Access needs
to be configured using either a short code, programmable button or source number.
• Intuity Emulation Mailbox Mode:For systems using Intuity emulation mode mailboxes,
the hunt group extension number and voicemail code can be used during normal mailbox
access.
• Avaya Branch Gateway Mailbox Mode or IP Office Mailbox Mode: For this mode of
mailbox access, short codes or a Voicemail Collect button are required to access the mailbox
directly.
The voicemail system (Voicemail Pro only) can be instructed to automatically forward messages
to the individual mailboxes of the hunt group members. The messages are not stored in the hunt
group mailbox.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Voicemail On Note:
From 11.1 FP1 IP Office system onwards, you can configure Voicemail On through
Group No Answer Destination under Group | Fallback tab.
Default = On
When on, the mailbox is used by the system to answer the any calls to the group that
reaches the Voicemail Answer Time. Note that selecting off does not disable the use of
the group mailbox. Messages can still be forward to the mailbox, and recordings can be
placed in it. The mailbox can also still be accessed to collect messages.
When a caller is directed to the voicemail to leave a message, the system indicates the
target user or hunt group mailbox.
• The mailbox of the originally targeted user or hunt group is used. This applies even if
the call was forwarded to another destination. It also includes scenarios where a hunt
group call overflows or is in fallback to another group.
• Voicemail Pro can be used to customize which mailbox is used separately from the
mailbox indicated by the system.
Voicemail Answer Note:
Time
From 11.1 FP1 IP Office system onwards, you can configure Voicemail Answer
Time through Group No Answer Time under Group | Fallback tab.
Default = 45 seconds. Range = Off, 1 to 99999 seconds.
This setting sets how long a call should be presented to a hunt group, and its overflow
groups if set, before going to the voicemail. When exceeded, the call goes to voicemail
(if available) regardless of any announcements, overflow, queuing, or any other actions.
If set to Off, voicemail is used when all available members of the hunt group have been
alerted for the no answer time.
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 Voicemail

Field Description
Voicemail Code Default = Blank. Range = 0 (no code) to 15 digits.
A code used by the voicemail server to validate access to this mailbox. If remote access
is attempted to a mailbox that has no voicemail code set, the prompt "Remote access is
not configured on this mailbox" is played.
The mailbox access code can be set through IP Office Manager or through the mailbox
telephone user interface (TUI). The minimum password length is:
• Voicemail Pro (Manager) - 0
• Voicemail Pro (Intuity TUI) - 2
• Embedded Voicemail (Manager) - 0
• Embedded Voicemail (Intuity TUI) - 0
Codes set through the Voicemail Pro telephone user interface are restricted to valid
sequences. For example, attempting to enter a code that matches the mailbox extension,
repeat the same number (1111) or a sequence of numbers (1234) are not allowed. If
these types of code are required they can be entered through Manager.
Manager does not enforce any password requirements for the code if one is set through
Manager.
• Embedded Voicemail For Embedded Voicemail running in IP Office mailbox mode, the
voicemail code is used if set.
• IP Office mode The voicemail code is required when accessing the mailbox from a
location that is not set as a trusted number in the user's Source Numbers list.
• Intuity Emulation mode By default the voicemail code is required for all mailbox
access. The first time the mailbox is accessed the user will be prompted to change the
password. Also if the voicemail code setting is left blank, the caller will be prompted to
set a code when they next access the mailbox. The requirement to enter the voicemail
code can be removed by adding a customized user or default collect call flow, refer to
the Voicemail Pro manuals for full details.
• Trusted Source Access The voicemail code is required when accessing the mailbox
from a location that is not set as a trusted number in the user's Source Numbers list.
• Call Flow Password Request Voicemail Pro call flows containing an action where the
action's PIN code set to $ will prompt the user for their voicemail code.
Voicemail Help Default = Off
This option controls whether users retrieving messages are automatically given an
additional prompt "For help at any time press 8." If switched off, users can still press
8 for help. For voicemail systems running in Intuity emulation mode, this option has no
effect. On those systems the default access greeting always includes the prompt "For
help at any time, press *4" (*H in the US locale).
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Group

Field Description
Broadcast Default = Off. (Voicemail Pro only).
When enabled, if a voicemail message is left for the hunt group, copies of the message
are forwarded to the mailboxes of the individual group members. The original message in
the hunt group mailbox is deleted unless it occurred as the result of call recording. This
feature is not applied to recordings created by Voice Question actions.
UMS Web Default = Off.
Services
This option is used with Voicemail Pro. If enabled, the hunt group mailbox can be
accessed using either an IMAP email client or a web browser. Note that the mailbox
must have a voicemail code set in order to use either of the UMS interfaces. UMS Web
Service licenses are required for the number of groups configured.
In the License section, double-clicking on the UMS Web Services license display a menu
that allows you to add and remove users and groups from the list of those enabled for
UMS Web Services without having to open the settings of each individual user or group.
Voicemail Email: Default = Blank (No voicemail email features)
This field is used to set the user or group email address used by the voicemail server for
voicemail email operation. When an address is entered, the additional Voicemail Email
control below are selectable to configure the type of voicemail email service that should
be provided.
Use of voicemail email requires the Voicemail Pro server to have been configured to
use either a local MAPI email client or an SMTP email server account. For Embedded
Voicemail, voicemail email is supportedand uses the system's SMTP settings.
The use of voicemail email for the sending (automatic or manual) of email messages
with wav files attached should be considered with care. A one-minute message creates a
1MB .wav file. Many email systems impose limits on emails and email attachment sizes.
For example the default limit on an Exchange server is 5MB.
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 Voicemail

Field Description
Voicemail Email Default = Off
If an email address is entered for the user or group, the following options become
selectable. These control the mode of automatic voicemail email operation provided by
the voicemail server whenever the voicemail mailbox receives a new voicemail message.
Users can change their voicemail email mode using visual voice. If the voicemail server
is set to IP Office mode, user can also change their voicemail email mode through the
telephone prompts. The ability to change the voicemail email mode can also be provided
by Voicemail Pro in a call flow using a Play Configuration Menu action or a Generic
action.
If the voicemail server is set to IP Office mode, users can manually forward a message to
email.
The options are:
• Off If off, none of the options below are used for automatic voicemail email. Users can
also select this mode by dialing *03 from their extension.
• Copy If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a copy of the message is attached to an email and sent to the
email address. There is no mailbox synchronization between the email and voicemail
mailboxes. For example reading and deletion of the email message does not affect the
message in the voicemail mailbox or the message waiting indication provided for that
new message.
• Forward If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, that message is attached to an email and sent to the email address.
No copy of the voicemail message is retained in the voicemail mailbox and their is no
message waiting indication. As with Copy, there is no mailbox synchronization between
the email and voicemail mailboxes. Users can also select this mode by dialing *01 from
their extension.
Note that until email forwarding is completed, the message is present in the voicemail
server mailbox and so may trigger features such as message waiting indication.
• UMS Exchange 2007 With Voicemail Pro, the system supports voicemail email to an
Exchange 2007 server email account. For users and groups also enabled for UMS
Web Services this significantly changes their mailbox operation. The Exchange Server
inbox is used as their voicemail message store and features such as message waiting
indication are set by new messages in that location rather than the voicemail mailbox
on the voicemail server. Telephone access to voicemail messages, including Visual
Voice access, is redirected to the Exchange 2007 mailbox.
• Alert If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a simple email message is sent to the email address. This is an
email message announcing details of the voicemail message but with no copy of the
voicemail message attached. Users can also select this mode by dialing *02 from their
extension.

Related links
Group on page 480

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Group

Voice Recording
Navigation: Group | Voice Recording
This tab is used to configure automatic recording of calls handled by hunt group members.
• Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
• Call recording starts when the call is answered.
• Call recording is paused when the call is parked or held. It restarts when the call is unparked
or taken off hold. This does not apply to SIP terminals.
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Recording continues for the duration of the call or up to the maximum recording time
configured on the voicemail server.
• Recording is stopped when the call ends or if:
- User call recording stops if the call is transferred to another user.
- Account code call recording stops if the call is transferred to another user.
- Hunt group call recording stops if the call is transferred to another user who is not a
member of the hunt group.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Record Inbound Default = None
Select whether automatic recording of incoming calls is enabled. The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. Otherwise, allow the call to continue without recording.
• Mandatory: Record the call if possible. Otherwise, block the call and return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording is always active.
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 Announcements

Field Description
Recording (Auto) Default = Mailbox
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This option set the destination for the recording to be a VRL
folder on the voicemail server. The VRL application polls that folder and collects waiting
recordings which it then places in its archive. Recording is still done by Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to the above but
instructs the voicemail server to create an authenticated recording. If the file contents
are changed, the file is invalidated though it can still be played.
- For systems recording to .opus format (the default), both settings create
authenticated recordings.
Auto Record Calls Default = External.
This setting allows selection of which calls are recorded. The options are External or
External & Internal.

Related links
Group on page 480

Announcements
Navigation: Group | Announcements
Announcements are played to callers waiting to be answered. This includes callers being
presented to hunt group members, ie. ringing, and callers queued for presentation.
• The system supports announcements using Voicemail Pro or Embedded Voicemail.
• If no voicemail channel is available for an announcement, the announcement is not played.
• In conjunction with Voicemail Pro, the system allows a number of voicemail channels to be
reserved for announcements. See System | Voicemail.
• With Voicemail Pro, the announcement can be replaced by the action specified in a Queued
(1st announcement) or Still Queued (2nd announcement) start point call flow. Refer to the
Voicemail Pro Installation and Maintenance documentation for details.
• Calls can be answered during the announcement. If it is a mandatory requirement that
announcements should be heard before a call is answered, then a Voicemail Pro call flow
should be used before the call is presented.
• A call becomes connected when the first announcement is played to it. That connected
state is signaled to the call provider who may start billing at that point. The call will also be
recorded as answered within the SMDR output once the first announcement is played.
• If a call is rerouted to a hunt group's Night Service Group or Out of Service Fallback Group,
the announcements of the new group are applied.

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Group

• If a call overflows, the announcements of the original group are still applied, not those of the
overflow group.
• For announcements to be used effectively, the hunt group's Voicemail Answer Time must
be extended or Voicemail On must be unselected.
Recording the Group Announcement
Voicemail Pro provides a default announcement "I'm afraid all the operators are busy but please
hold and you will be transferred when somebody becomes available". This default is used
for announcement 1 and announcement 2 if no specific hunt group announcement has been
recorded. Embedded Voicemail does not provide any default announcement. Voicemail Lite also
provides the default announcements.
The maximum length for announcements is 10 minutes. New announcements can be recorded
using the following methods.
Voicemail Lite: Access the hunt group mailbox and press 3. Then press either 3 to record the 1st
announcement for the hunt group or 4 to record the 2nd announcement for the hunt group.
Voicemail Pro : The method of recording announcements depends on the mailbox mode being
used by the voicemail server.
• IP Office Mailbox Mode: Access the hunt group mailbox and press 3. Then press either 3 to
record the 1st announcement for the hunt group or 4 to record the 2nd announcement for the
hunt group.
• Intuity Emulation Mailbox Mode: There is no mechanism within the Intuity telephony user
interface (TUI) to record hunt group announcements. To provide custom announcements,
hunt group queued and still queued start points must be configured with Voicemail Pro with
the required prompts played by a generic action.
Embedded Voicemail: Embedded Voicemail does not include any default announcement or
method for recording announcements. The Record Message short code feature is provided to
allow the recording of announcements. The telephone number field of short codes using this
feature requires the extension number followed by either ".1" for announcement 1 or ".2" for
announcement 2. For example, for extension number 300, the short codes *91N# | Record
Message | N".1" and *92N# | Record Message | N".2" could be used to allow recording of the
announcements by dialing *91300# and *92300#.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Announcements Default = Off.
On
This setting enables or disables announcements.
Wait before 1st Default = 10 seconds. Range = 0 to 255 seconds.
announcement:
This setting sets the time delay from the calls presentation, after which the first
announcement should be played to the caller. If Synchronize Calls is selected, the actual
wait may differ, see below.
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 Announcements

Field Description
Flag call as Default = Off.
answered
This setting is used by the CCC and CBC applications. By default they do not regarded
a call as answered until it has been answered by a person or by a Voicemail Pro action
with Flag call as answered selected. This setting allows calls to be marked as answered
once the caller has heard the first announcement.
Post Default = Music on hold.
announcement
Following the first announcement, you can select whether the caller should hear Music on
tone
Hold, Ringing or Silence until answered or played another announcement.
2nd Default = On.
Announcement
If selected, a second announcement can be played to the caller if they have still not been
answered.
Wait before 2nd Default = 20 seconds. Range = 0 to 255 seconds.
announcement
This setting sets the wait between the 1st and the 2nd announcement. If Synchronize
Calls is selected, the actual wait may differ, see below.
Repeat last Default = On.
announcement
If selected, the last announcement played to the caller is repeated until they are answered
or hang-up.
Wait before repeat Default = 20 seconds. Range = 0 to 255 seconds.
If Repeat last announcement is selected, this setting sets is applied between each
repeat of the last announcement. If Synchronize Calls is selected, this value is grayed
out and set to match the Wait before 2nd announcement setting.
Synchronize calls Default = Off
This option can be used to reduce the number of voicemail channels required to provide
the announcements. Using this setting, the maximum number of voicemail channels
needed is 1 or 2 depending on the number of selected announcements.
• When on:
- If the required prompt is already being played to another caller, further callers wait until
the prompt has completed and can be restarted.
- If the required prompt is not being played and there are multiple waiting callers, once
one caller has waited for the set wait period, the prompt is played to all the currently
waiting callers.
- If Voicemail Pro custom Queued or Still Queued start point call flows are used for
the announcements, when Synchronize Calls is enabled the call flows support the
playing of prompts only.
• When off:
- Announcements are played individually for each call. This requires a separate
voicemail channel each time an announcement is played to each caller. While this
accurately follows the wait settings, it does not use voicemail channels efficiently.

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Group

Related links
Group on page 480

SIP
Navigation: Group | SIP
Each hunt group can be configured with its own SIP URI information. For calls received on a SIP
line where any of the line's SIP URI fields are set to Use Internal Data, if the call is presented to
the hunt group that data is taken from these settings.
This form is hidden if there are no system multi-site network lines in the configuration or no SIP
lines with a URI set to Use Internal Data.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
SIP Name Default = Blank on Voicemail tab/Extension number on other tabs.
This value is used for fields, other the Contact header, where the SIP URI entry being
used has its Contact field set to Use Internal Data.
• On incoming calls, if the Local URI is set to Use Internal Data, the system can
potentially match the received R-URI or From header value to a user and/or group SIP
Name. This requires the SIP URIs Incoming Group to match a Incoming Call Route
with the same Line Group ID and a . (period) destination.
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display field of the SIP URI being used is set to
Use Internal Data.
Contact Default = Blank on Voicemail tab/Extension number on other tabs.
The value is used for the Contact header when the Contact field of the SIP URI being
used for a SIP call is set to Use Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs.
If the From field in the SIP URI is set to Use Internal Data, selecting this option inserts
Anonymous into that field rather than the SIP Name set above. See Anonymous SIP
Calls on page 842.

Related links
Group on page 480

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Chapter 31: Short Code

Navigation: Short Code | Short Code

Configuration settings
These settings are used to create System Short Codes. System short codes can be dialed by all
system users. However the system short code is ignored if the user dialing matches a user or user
rights short code.
Warning:
User dialing of emergency numbers must not be blocked. If short codes are edited, the users
ability to dial emergency numbers must be tested and maintained.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
• For systems using record consolidation, you can only add and edit this type of record at the
solution level. The record is then automatically copied to each IP Office system in the network.
Field Description
Code The dialing digits used to trigger the short code. Maximum length 31 characters. For
details of the characters that you can use, see Short Code Characters on page 935.
Feature Select the action to be performed by the short code. For descriptions of the features, see
Short Code Features on page 953.
Telephone The number dialed by the short code or parameters for the short code feature. This field
Number can contain numbers and characters. For example, it can contain Voicemail Pro start
point names, user names, hunt group names and telephone numbers (including those
with special characters). Maximum length 31 characters. See Short Code Characters on
page 935.
The majority of North-American telephony services and SIP trunks use 'en-bloc' dialing.
That is, they expect to receive the routing digits for a call as a single simultaneous set.
Therefore, the use of a ; is recommended at the end of all dialing short codes that use
an N. This is also recommended for all dialing where secondary dial tone short codes are
being used.
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Short Code

Field Description
Line Group ID Default = 0.
For short codes that result in the dialing of a number, that is short codes with a Dial
feature, this field is used to enter the initially routing destination of the call. The drop down
can be used to select the following from the displayed list:
• Outgoing Group ID: The Outgoing Group ID's current setup within the system
configuration are listed. If an Outgoing Group ID is selected, the call will be routed
to the first available line or channel within that group.
• ARS: The ARS records currently configured in the system are listed. If an ARS record
is selected, the call will be routed by the setting within that ARS record. Refer to ARS
Overview.
• For calls matching Dial Emergency short codes, this setting is overridden by the
Emergency ARS settings of the dialing extension's location.
Locale Default = Blank.
For short codes that route calls to voicemail, this field can be used to set the prompts
locale that should be used if available on the voicemail server.
Force Account Default = Off.
Code
For short codes that result in the dialing of a number, this field trigger the user being
prompted to enter a valid account code before the call is allowed to continue.
Force Default = Off.
Authorization
This option is only shown on systems where authorization codes have been enabled. If
Code
selected, then for short codes that result in the dialing of a number, the user is required to
enter a valid authorization code in order to continue the call.

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Chapter 32: Services

Navigation Path: Services

Services are used to configure the settings required when a user or device on the LAN needs to
connect to a off-switch data service such as the Internet or another network. Services can be used
when making data connections via trunk or WAN interfaces.
After creating a service, it can be used as the destination for an IP Route record. One service can
also be set as the Default Service. That service will then be used for any data traffic received by
the system for which no IP Route is specified.
The system supports the following types of service:
Service Description
Remote Support This type of tunnel is used by subscription mode IP Office systems for RSS connections
Services routed to the system through COM. For details, refer to Using Customer Operations
Manager for IP Office Subscription Systems.
Normal Service This type of service should be selected when for example, connecting to an ISP.
WAN Service This type of service is used when creating a WAN link. A User and RAS Service will
also be created with the same name. These three records are automatically linked and
each open the same form. Note however, that this type of Service cannot be used if the
Encrypted Password option is checked. In this case, the RAS Service name must match
the Account Name. Therefore either create each record manually or create an Intranet
Service.
Intranet Service This type of service can be selected to automatically create a User with the same name
at the same time. These two records are linked and will each open the same form. The
User's password is entered in the Incoming Password field at the bottom on the Service
tab. An Intranet Services shares the same configuration tabs as those available to the
WAN Service.
SSL VPN The SSL VPN service provides secure tunneling between the Avaya IP Office hardware
installed at a customer site and a remote Avaya VPN Gateway (AVG). This secure
tunnel allows support personnel to offer remote management services to customers,
such as fault management, monitoring, and administration. Refer to the Deploying Avaya
IP Office™ Platform SSL VPN Services manual.

Related links
Remote Support Services on page 506
Service on page 506
Bandwidth on page 507
IP on page 510

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Services

Autoconnect on page 511

Quota on page 511
PPP on page 512
Fallback on page 514
Dial In on page 515
SSL VPN Service on page 515

Remote Support Services

Navigation: Services > Remote Support Services
This type of tunnel is used by subscription mode IP Office systems for RSS connections routed to
the system through COM. For details, refer to Using Customer Operations Manager for IP Office
Subscription Systems.
TCP Tunnels
These settings are used to configure allowed TCP tunnel connections.
Field Description
Application Default = Blank
You can use the drop-down menu to select from a range of services (OneXPortal,
SSH, Telnet, RDP, WebControl). The Protocol and Server Port Number fields are
then pre-filled with the defaults for the selected application. For other services, select
Custom.
Protocol Default = Blank
Only TCP is supported.
Server IP Address Default = Blank
The address of the server to which the RSS tunnel connects.
Server Port Default = Blank
Number
The server port for the tunnel connection.

Related links
Services on page 505

Service
Navigation: Service | Service

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 Bandwidth

Additional configuration information

This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Configuration settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Service Name The name of the service. It is recommended that only alphanumeric characters be used.
Account Name The user name that is used to authenticate the connection. This is provided by the ISP or
remote system.
Password Default = Blank
Enter the password that is used to authenticate the connection. This is provided by the
ISP or remote system.
Telephone Number Default = Blank
If the connection is to be made via ISDN enter the telephone number to be dialed. This is
provided by the ISP or remote system.
Firewall Profile Default = Internet01 if present, otherwise <None>
From the list box select the Firewall Profile that is used to allow/disallow protocols
through this Service.
Encrypted Default = Off When enabled the password is authenticated via CHAP (this must also be
Password supported at the remote end). If disabled, PAP is used as the authentication method.
Default Route Default = Off
When enabled this Service is the default route for data packets unless a blank IP Route
has been defined in the system IP Routes. A green arrow appears to the left of the
Service in the Configuration Tree. Only one Service can be the default route. If disabled,
a route must be created under IP Route.
Incoming Default = Blank Shown on WAN and Intranet services. Enter the password that will
Password be used to authenticate the connection from the remote Control Unit. (If this field has
appeared because you have created a Service and User of the same name, this is the
password you entered in the User's Password field).

Related links
Services on page 505

Bandwidth
Navigation: Service | Bandwidth
These options give the ability to make ISDN calls between sites only when there is data to be sent
or sufficient data to warrant an additional call. The calls are made automatically without the users
being aware of when calls begin or end. Using ISDN it is possible to establish a data call and be
passing data in less that a second.

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Services

Note:
The system will check Minimum Call Time first, then Idle Period, then the Active Idle
Period.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Minimum No of Default = 1. Range = 1 to 30.
Channels
Defines the number of channels used to connect for an outgoing connection. The initial
channel must be established and stable, before further calls are made.
Maximum No of Default = 1. Range = 1 to 30.
Channels
Defines the maximum number of channels to can be used. This field should contain a
value equal to or greater than the Minimum Channels field.
Extra BW Default = 50%. Range = 0 to 100%.
Threshold
Defines the utilization threshold at which extra channels are connected. The value
entered is a %. The % utilization is calculated over the total number of channels in
use at any time, which may be one, two etc.
For example, if Minimum Channels set to 1, Maximum Channels set to 2 and Extra
Bandwidth set to 50 - once 50% of first channel has been used the second channel is
connected.
Reduce BW Default = 10%. Range = 0 to 100%.
Threshold
Defines the utilization threshold at which additional channels are disconnected. The
value entered is a %. Additional calls are only dropped when the % utilization, calculated
over the total number of channels in use, falls below the % value set for a time period
defined by the Service-Idle Time. The last call (calls - if Minimum Calls is greater than 1)
to the Service is only dropped if the % utilization falls to 0, for a time period defined by
the Service-Idle Time. Only used when 2 or more channels are set above.
For example, if Minimum Channels set to 1, Maximum Channels set to 2 and Reduce
Bandwidth is set to 10 - once the usage of the 2 channels drops to 10% the number of
channels used is 1.
Callback Default = Blank
Telephone Number
The number that is given to the remote service, via BAP, which the remote Control
Unit then dials to allow the bandwidth to be increased. Incoming Call routing and RAS
Services must be appropriately configured.
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 Bandwidth

Field Description
Idle Period (secs) Default = 10 seconds. Range = 0 to 999999 seconds.
The time period, in seconds, required to expire after the line has gone idle. At this point
the call is considered inactive and is completely closed.
For example, the 'Idle Period' is set to X seconds. X seconds before the 'Active Idle
Period' timeouts the Control Unit checks the packets being transmitted/received, if there
is nothing then at the end of the 'Active Idle Period' the session is closed & the line is
dropped. If there are some packets being transmitted or received then the line stays up.
After the 'Active Idle Period' has timed out the system performs the same check every X
seconds, until there are no packets being transferred and the session is closed and the
line dropped.
Active Idle Period Default = 180 seconds. Range = 0 to 999999 seconds.
(secs):
Sets the time period during which time the line has gone idle but there are still active
sessions in progress (for example an FTP is in process, but not actually passing data at
the moment). Only after this timeout will call be dropped.
For example, you are downloading a file from your PC and for some reason the other
end has stopped responding, (the remote site may have a problem etc.) the line is idle,
not down, no data is being transmitted/ received but the file download session is still
active. After the set time period of being in this state the line will drop and the sessions
close. You may receive a remote server timeout error on your PC in the Browser/FTP
client you were using.
Minimum Call Time Default = 60 seconds. Range = 0 to 999999 seconds.
(secs):
Sets the minimum time that a call is held up after initial connection. This is useful if you
pay a minimum call charge every time a call is made, no matter the actual length of the
call. The minimum call time should be set to match that provided by the line provider.
Extra Bandwidth Default = Incoming Outgoing
Mode
Defines the mode of operation used to increases bandwidth to the initial call to the
remote Service. The options are:
• Outgoing Only Bandwidth is added by making outgoing calls.
• Incoming Only Bandwidth is added by the remote service calling back on the BACP
number (assuming that BACP is successfully negotiated).
• Outgoing Incoming Uses both methods but bandwidth is first added using outgoing
calls.
• Incoming Outgoing Uses both methods but bandwidth is first added using incoming
BACP calls.

Related links
Services on page 505

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Services

IP
Navigation: Service | IP
The fields in this tab are used to configure network addressing for the services you are running.
Depending on how your network is configured, the use of Network Address Translation (NAT) may
be required.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
IP Address Default = 0.0.0.0 (address assigned by ISP)
An address should only be entered here if a specific IP address and mask have been
provided by the Service Provider. Note that if the address is in a different domain from the
system then NAT is automatically enabled
IP Mask Default = 0.0.0.0 (use NAT)
Enter the IP Mask associated with the IP Address if an address is entered.
Primary Transfer Default = 0.0.0.0 (No transfer)
IP Address
This address acts as a primary address for incoming IP traffic. All incoming IP packets
without a session are translated to this address. This would normally be set to the local
mail or web server address.
For control units supporting a LAN1 and LAN2, the primary transfer address for each LAN
can be set through the System | LAN1 and System | LAN2 tabs.
RIP Mode Default = None
Routing Information Protocol (RIP) is a method by which network routers can exchange
information about device locations and routes. RIP can be used within small networks to
allow dynamic route configuration as opposed to static configuration using. The options
are:
• None The LAN does not listen to or send RIP messages.
• Listen Only (Passive) Listen to RIP-1 and RIP-2 messages in order to learn RIP routes
on the network.
• RIP1 Listen to RIP-1 and RIP-2 messages and send RIP-1 responses as a sub-network
broadcast.
• RIP2 Broadcast (RIP1 Compatibility) Listen to RIP-1 and RIP-2 messages and send
RIP-2 responses as a sub-network broadcast.
• RIP2 Multicast Listen to RIP-1 and RIP-2 messages and send RIP-2 responses to the
RIP-2 multicast address.
Request DNS Default = Off.
When selected, DNS information is obtained from the service provider. To use this, the
DNS Server addresses set in the system configuration (System | DNS) should be blank.
The PC making the DNS request should have the system set as its DNS Server. For
DHCP clients the system will provide its own address as the DNS server.
Table continues…

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 Autoconnect

Field Description
Forward Multicast Default = On.
Messages
By default this option is on. Multicasting allows WAN bandwidth to be maximized through
the reduction of traffic that needs to be passed between sites.

Related links
Services on page 505

Autoconnect
Navigation: Service | Autoconnect
These settings enable you to set up automatic connections to the specified Service.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Auto Connect Default = 0 (disabled). Range = 0 to 99999 minutes.
Interval (mins):
This field defines how often this Service will automatically be called ("polled"). For
example setting 60 means the system will call this Service every hour in the absence
of any normally generated call (this timer is reset for every call; therefore if the service is
already connected, then no additional calls are made). This is ideal for SMTP Mail polling
from Internet Service Providers.
Auto Connect Default = <None>
Time Profile
Allows the selection of any configured Time Profiles. The selected profile controls the time
period during which automatic connections to the service are made. It does NOT mean
that connection to that service is barred outside of these hours. For example, if a time
profile called "Working Hours" is selected, where the profile is defined to be 9:00AM to
6:00PM Monday to Friday, then automatic connection to the service will not be made
unless its within the defined profile. If there is an existing connection to the service at
9:00AM, then the connection will continue. If there is no connection, then an automatic
connection will be made at 9:00AM.

Related links
Services on page 505

Quota
Navigation: Service | Quota
Quotas are associated with outgoing calls, they place a time limit on calls to a particular IP
Service. This avoids excessive call charges when perhaps something changes on your network
and call frequency increases unintentionally.

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Services

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Quota Time Default = 240 minutes. Range = 0 to 99999 minutes.
(mins)
Defines the number of minutes used in the quota. When the quota time is used up no
further data can be passed to this service. This feature is useful to stop things like an
internet game keeping a call to your ISP open for a long period.

Warning:
Setting a value here without selecting a Quota period below will stop all further calls
after the Quota Time has expired.
Quota: Default = Daily. Range = None, Daily, Weekly or Monthly
Sets the period during which the quota is applied. For example, if the Quota Time is 60
minutes and the Quota is set to Daily, then the maximum total connect time during any
day is 60 minutes. Any time beyond this will cause the system to close the service and
prevent any further calls to this service. To disable quotas select None and set a Quota
Time of zero.

Note:
The ClearQuota feature can be used to create short codes to refresh the quota time.

Related links
Services on page 505

PPP
Navigation: Service | PPP
These settings enable you to configure Point to Point Protocol (PPP) in relation to this particular
service. PPP is a protocol for communication between two computers using a Serial interface.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Chap Challenge Default = 0 (disabled). Range = 0 to 99999 seconds. The period between CHAP
Interval (secs) challenges. Blank or 0 disables repeated challenges.
Bi-Directional Default =Off.
Chap
Header Default = None selected
Compression
Enables the negotiation and use of IP Header Compression. Supported modes are IPHC
and VJ. IPHC should be used on WAN links.
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 PPP

Field Description
PPP Compression Default = MPPC
Mode
Enables the negotiate and use of compression. Do not use on VoIP WAN links. The
options are:
• Disable Do not use or attempt to use compression.
• StacLZS Attempt to use STAC compression (Mode 3, sequence check mode).
• MPPC Attempt to use MPPC compression. Useful for NT Servers.
PPP Callback Default = Disabled.
Mode
The options are:
• Disable Callback is not enabled
• LCP (Link Control Protocol) After authentication the incoming call is dropped and an
outgoing call to the number configured in the Service is made to re-establish the link.
• Callback CP (Microsoft's Callback Control Protocol) After acceptance from both ends
the incoming call is dropped and an outgoing call to the number configured in the
Service is made to re-establish the link.
• Extended CBCP (Extended Callback Control Protocol) Similar to Callback CP except
the Microsoft application at the remote end prompts for a telephone number. An
outgoing call is then made to that number to re-establish the link.
PPP Access Mode Default = Digital64
Sets the protocol, line speed and connection request type used when making outgoing
calls. Incoming calls are automatically handled (see RAS services). The options are:
• Digital64 Protocol set to Sync PPP, rate 64000 bps, call presented to local exchange as
a "Data Call".
• Digital56 As above but rate 56000 bps.
• Voice56 As above but call is presented to local exchange as a "Voice Call".
• V120 Protocol set to Async PPP, rate V.120, call presented to local exchange as a "Data
Call". This mode runs at up to 64K per channel but has a higher Protocol overhead than
pure 64K operation. Used for some bulletin board systems as it allows the destination
end to run at a different asynchronous speed to the calling end.
• V110 Protocol is set to Async PPP, rate V.110. This runs at 9600 bps, call is presented
to local exchange as a "Data Call". It is ideal for some bulletin boards.
• Modem Allows Asynchronous PPP to run over an auto-adapting Modem to a service
provider (requires a Modem2 card in the main unit)
Data Pkt. Size Default = 0. Range = 0 to 2048.
Sets the size limit for the Maximum Transmissible Unit.
BACP Default = Off.
Enables the negotiation and use of BACP/BCP protocols. These are used to control the
addition of B channels to increase bandwidth.
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Services

Field Description
Incoming traffic Default = On.
does not keep link
When enabled, the link is not kept up for incoming traffic only.
up
Multilink/QoS Default = Off.
Enables the negotiation and use of Multilink protocol (MPPC) on links into this Service.
Multilink must be enabled if there is more than one channel that is allowed to be Bundled/
Multilinked to this RAS Service.

Related links
Services on page 505

Fallback
Navigation: Service | Fallback
These settings allow you to set up a fallback for the Service. For example, you may wish to
connect to your ISP during working hours and at other times take advantage of varying call
charges from an alternative carrier. You could therefore set up one Service to connect during peak
times and another to act as fallback during the cheaper period.
You need to create an additional Service to be used during the cheaper period and select this
service from the Fallback Service list box (open the Service form and select the Fallback tab).
If the original Service is to be used during specific hours and the Fallback Service to be used
outside of these hours, a Time Profile can be created. Select this Time Profile from the Time
Profile list box. At the set time the original Service goes into Fallback and the Fallback Service is
used.
A Service can also be put into Fallback manually using short codes, for example:
Put the service "Internet" into fallback:
• Short Code: *85
• Telephone Number: "Internet"
• Line Group ID: 0
• Feature: SetHuntGroupNightService
Take the service "Internet" out of fallback:
• Short Code: *86
• Telephone Number: "Internet"
• Line Group ID: 0
• Feature: ClearHuntGroupNightService
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Dial In

Field Description
In Fallback Default = Off.
This option indicates whether the Service is in Fallback or not. A service can be set into
fallback using this setting. Alternatively a service can be set into fallback using a time
profile or short codes.
Time profile Default = <None> (No automatic fallback)
Select the time profile you wish to use for the service. The time profile should be set up
for the hours that you wish this service to be operational, out of these hours the Fallback
Service is used.
Fallback Service Default = <None>
Select the service that is used when this service is in fallback.

Related links
Services on page 505

Dial In
Navigation: Service | Dial In
Only available for WAN and Intranet Services. This tab is used to define a WAN connection.
To define a WAN connection, click Add and enter WAN if the service is being routed via a WAN port
on a WAN3 expansion module.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Related links
Services on page 505

SSL VPN Service

The SSL VPN service provides secure tunneling between the Avaya IP Office hardware installed
at a customer site and a remote Avaya VPN Gateway (AVG). This secure tunnel allows support
personnel to offer remote management services to customers, such as fault management,
monitoring, and administration.
For full details on how to configure and administer SSL VPN services, refer to the Deploying
Avaya IP Office™ Platform SSL VPN Services manual.
Related links
Services on page 505
Service on page 516
Session on page 516

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Services

NAPT on page 517

Fallback on page 518

Service
Navigation: Service | SSL VPN Service | Service
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Service Name Enter a name for the SSL VPN service.
Account Name Enter the SSL VPN service account name. This account name is used for authenticating
the SSL VPN service when connecting with the Avaya VPN Gateway (AVG).
Account Password Enter the password for the SSL VPN service account.
Confirm Password Confirm the password for the SSL VPN service account.
Server Address Enter the address of the VPN gateway. The address can be a fully qualified domain
name or an IPv4 address
Server Type Default = AVG. This field is fixed to AVG (Avaya VPN Gateway).
Server Port Default = 443. Select a port number.
Number

Related links
SSL VPN Service on page 515

Session
Navigation: Service | SSL VPN Service | Session
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Session Mode Default = Always On.
This setting is greyed out and cannot be adjusted.
Preferred Data Default = UDP.
Transport Protocol
This is the protocol used by the SSL VPN service for data transport. Only TCP is
supported. If you select UDP as the protocol when you configure the connection, UDP
displays in this field but the SSL VPN service falls back to TCP.
Heartbeat Interval Default = 30 seconds. Range = 1 to 600 seconds.
Enter the length of the interval between heartbeat messages, in seconds. The default
value is 30 seconds.
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 SSL VPN Service

Field Description
Heartbeat Retries Default = 4. Range = 1 to 10.
Enter the number of unacknowledged heartbeat messages that IP Office sends to
AVG before determining that AVG is not responsive. When this number of consecutive
heartbeat messages is reached and AVG has not acknowledged them, IP Office ends
the connection.
Keepalive Interval Default = 10 seconds. Range = 0 (Disabled) to 600 seconds.
Not used for TCP connections. Keepalive messages are sent over the UDP data
transport channel to prevent sessions in network routers from timing out.
Reconnection Default = 60 seconds. Range = 1 to 600 seconds.
Interval on Failure
The interval the system waits attempting to re-establish a connection with the AVG.
The interval begins when the SSL VPN tunnel is in-service and makes an unsuccessful
attempt to connect with the AVG, or when the connection with the AVG is lost. The
default is 60 seconds.

Related links
SSL VPN Service on page 515

NAPT
Navigation: Service | SSL VPN Service | NAPT
The Network Address Port Translation (NAPT) rules are part of SSL VPN configuration. NAPT
rules allow a support service provider to remotely access LAN devices located on a private IP
Office network. You can configure each SSL VPN service instance with a unique set of NAPT
rules. You can configure up to 64 rules.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
The SSL VPN restarts after a setting change.
Field Description
Application Default = Blank
Defines the communication application used to connect to the LAN device through the
SSL VPN tunnel. When you select an application, the Protocol and Port Number fields
are filled with the default values. The drop-down Application selector options and the
associated default values are:
External and Internal Port
Application Protocol Number
Custom TCP 0
VMPro TCP 50791
OneXPortal TCP 8080
SSH TCP 22
TELNET TCP 23
Table continues…

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Services

Field Description
RDP TCP 3389
WebControl TCP 7070
Protocol Default = TCP
The protocol used by the application. The options are TCP and UDP.
External Port Default = the default port number for the application. Range = 0 to 65535
Number
Defines the port number used by the application to connect from the external network to
the LAN device in the customer private network.
Internal IP Default = Blank.
address
The IP address of the LAN device in the customer network.
Internal Port Default = the default port number for the application. Range = 0 to 65535
Number
Defines the port number used by the application to connect to the LAN device in the
customer private network.

Related links
SSL VPN Service on page 515

Fallback
Navigation: Service | SSL VPN Service | Fallback
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
In Fallback Default = Off.
This setting is used to indicate whether the SSL VPN service is in use or not.
• To configure the service without establishing an SSL VPN connection, or to disable an
SSL VPN connection, select this option.
• To enable the service and establish an SSL VPN connection, de-select this option.
• The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively.
The service is indicated by setting the service name as the telephone number or action
data. Do not use quotation marks.

Related links
SSL VPN Service on page 515

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Chapter 33: RAS

Navigation: RAS | RAS

A Remote Access Server (RAS) is a piece of computer hardware which sits on a corporate LAN and
into which employees dial on the public switched telephone network to get access to their email and
to software and data on the corporate LAN.
This form is used to create a RAS service that the system offers Dial In users. A RAS service
is needed when configuring modem dial in access, digital (ISDN) dial in access and a WAN link.
Some systems may only require one RAS service since the incoming call type can be automatically
sensed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Name A textual name for this service. If Encrypted Password below is used, this name must
match the Account Name entered in the Service form.
Extension Enter an extension number if this service is to be accessed internally.
COM Port For future use.
TA Enable Default = Off
Select to enable or disable - if enabled RAS will pass the call onto a TA port for external
handling.
Encrypted Default = Off
Password
This option is used to define whether Dial In users are asked to use PAP or CHAP during
their initial log in to the RAS Service. If the Encrypted Password box is checked then Dial
In users are sent a CHAP challenge, if the box is unchecked PAP is used as the Dial In
Authorization method.

Related links
PPP on page 519

PPP
Navigation: RAS | PPP
PPP (Point-to-Point Protocol) is a Protocol for communication between two computers using a
Serial interface, typically a personal computer connected by phone line to a server.

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RAS

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
CHAP Challenge Default = 0 (disabled). Range = 0 to 99999 seconds.
Interval (secs)
The period between successive CHAP challenges. Blank or 0 disables repeated
challenges.
Header Default = Off
Compression
Enables the negotiation and use of IP Header Compression as per RFC2507, RFC2508
and RFC2509.
PPP Compression Default = MPPC This option is used to negotiate compression (or not) using CCP. If set to
Mode MPPC or StacLZS the system will try to negotiate this mode with the remote Control Unit.
If set to Disable CCP is not negotiated. The options are:
• Disable Do not use or attempt to use compression.
• StacLZS Attempt to use and negotiate STAC compression (the standard, Mode 3)
• MPPC Attempt to use and negotiate MPPC (Microsoft) compression. Useful for dialing
into NT Servers.
PPP Callback Default = Disable
Mode
The options are:
• Disable: Callback is not enabled
• LCP: (Link Control Protocol) After authentication the incoming call is dropped and an
outgoing call to the number configured in the Service will be made to reestablish the
link.
• Callback CP: (Microsoft's Callback Control Protocol) After acceptance from both ends
the incoming call is dropped and an outgoing call to the number configured in the
Service is made to reestablish the link.
• Extended CBCP: (Extended Callback Control Protocol) Similar to Callback CP however
the Microsoft application at the remote end will prompt for a telephone number. An
outgoing call will then be made to that number to reestablish the link.
Data Pkt. Size Default = 0. Range = 0 to 2048.
This is the number of data bytes contained in a Data Packet.
BACP Default = Off
Allows negotiation of the BACP/BCP protocols. These are used to control the addition of
additional B channels to simultaneously improve data throughput.
Multilink Default = Off
When enabled the system attempts to negotiate the use of the Multilink protocol (MPPC)
on the link(s) into this Service. Multilink must be enabled if the more than one channel is
allowed to be Bundled/Multilinked to this RAS Service.

Related links
RAS on page 519

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Chapter 34: Incoming Call Route

Incoming call routes are used to determine the destination of voice and data calls received by the
system. On systems where a large number incoming call routes need to be setup for DID numbers,
the MSN/DID Configuration tool can be used.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can be
used to modify the incoming digits.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Determining which incoming call route is used is based on the call matching a number of possible
criteria. In order of highest priority first, the criteria, which if set must be matched by the call in order
for the call to use that route are:
1. The Bearer Capability indicated, if any, with the call. For example whether the call is a
voice, data or video call.
2. The Incoming Group ID of the trunk or trunk channel on which the call was received.
3. The Incoming Number received with the call.
4. The Incoming Sub Address received with the call.
5. The Incoming CLI of the caller.
Multiple Matches
If there is a match between more than one incoming call route record, the one added to the
configuration first is used.
Incoming Call Route Destinations
Each incoming route can include a fallback destination for when the primary destination is busy. It
can also include a time profile which control when the primary destination is used. Outside the time
profile calls are redirected to a night service destination. Multiple time profiles can be associated
with an incoming call route. Each time profile used has its own destination and fallback destination
specified.
Incoming Call Routing Examples
Example 1
For this example, the customer has subscribes to receive two 2-digit DID numbers. They want calls
on one routed to a Sales hunt group and calls on the other to a Services hunt group. Other calls
should use the normal default route to hunt group Main. The following incoming call routes were
added to the configuration to achieve this:

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Incoming Call Route

Line Group Incoming Number Destination

0 77 Sales
0 88 Services
0 blank Main

Note that the incoming numbers could have been entered as the full dialed number, for example
7325551177 and 7325551188 respectively. The result would still remain the same as incoming
number matching is done from right-to-left.
Line Group Incoming Number Destination
0 7325551177 Sales
0 7325551188 Services
0 blank Main

Example 2
In the example below the incoming number digits 77 are received. The incoming call route records
677 and 77 have the same number of matching digit place and no non-matching places so both a
potential matches. In this scenario the system will use the incoming call route with the Incoming
Number specified for matching.
Line Group Incoming Number Destination
0 677 Support
0 77 Sales
0 7 Services
0 blank Main

Example 3
In the following example, the 677 record is used as the match for 77 as it has more matching digits
than the 7 record and no non-matching digits.
Line Group Incoming Number Destination
0 677 Support
0 7 Services
0 blank Main

Example 4
In this example the digits 777 are received. The 677 record had a non-matching digit, so it is not a
match. The 7 record is used as it has one matching digit and no non-matching digits.
Line Group Incoming Number Destination
0 677 Support
0 7 Services
0 blank Main

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 Standard

Example 5
In this example the digits 77 are received. Both the additional incoming call routes are potential
matches. In this case the route with the shorter Incoming Number specified for matching is used and
the call is routed to Services.
Line Group Incoming Number Destination
0 98XXX Support
0 8XXX Services
0 blank Main

Example 6
In this example two incoming call routes have been added, one for incoming number 6XXX and one
for incoming number 8XXX. In this case, any three digit incoming numbers will potential match both
routes. When this occurs, potential match that was added to the system configuration first is used. If
4 or more digits were received then an exact matching or non-matching would occur.
Line Group Incoming Number Destination
0 6XXX Support
0 8XXX Services
0 blank Main

Related links
Standard on page 523
Voice Recording on page 527
Destinations on page 528

Standard
Navigation: Incoming Call Route | Standard
Additional configuration information
For additional information on the Tag setting, see Call Tagging on page 719.
Incoming call routes are used to match call received with destinations. Routes can be based
on the incoming line group, the type of call, incoming digits or the caller's ICLID. If a range of
MSN/DID numbers has been issued, this form can be populated using the MSN Configuration tool.
In Manager, see Tools > MSN Configuration.
Default Blank Call Routes
By default the configuration contains two incoming calls routes; one set for Any Voice calls
(including analog modem) and one for Any Data calls. While the destination of these default
routes can be changed, it is strongly recommended that the default routes are not deleted.
• Deleting the default call routes, may cause busy tone to be returned to any incoming external
call that does not match any incoming call route.

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Incoming Call Route

• Setting any route to a blank destination field, may cause the incoming number to be checked
against system short codes for a match. This may lead to the call being rerouted off-switch.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
If there is no matching incoming call route for a call, matching is attempted against system short
codes and finally against voicemail nodes before the call is dropped.
SIP Calls
For SIP calls, the following fields are used for call matching:
• Line Group ID This field is matched against the Incoming Group settings of the SIP URI
(Line | SIP URI). This must be an exact match.
• Incoming Number This field can be used to match the called details (TO) in the SIP header
of incoming calls. It can contain a number, SIP URI or Tel URI. For SIP URI's the domain part
of the URI is removed before matching by incoming call routing occurs. For example, for the
SIP URI [email protected] , only the user part of the URI, ie. mysip, is used for matching.
The Call Routing Method setting of the SIP line can be used to select whether the value used for
incoming number matching is taken from the To Header or the Request URI information provided
with incoming calls on that line.
Incoming CLI This field can be used to match the calling details (FROM) in the SDP header of
incoming SIP calls. It can contain a number, SIP URI, Tel URI or IP address received with SIP
calls. For all types of incoming CLI except IP addresses a partial record can be used to achieve
the match, records being read from left to right. For IP addresses only full record matching is
supported.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Incoming Call Matching Fields:
The following fields are used to determine if the Incoming Call Route is a potential match for the
incoming call. By default the fields are used for matching in the order shown starting with Bearer
Capability.
Field Description
Line Group ID Default = 0. Range = 0 to 99999.
Matches against the Incoming Line Group to which the trunk receiving the call belongs.
For Server Edition systems, the default value 0 is not allowed. You must change the
default value and enter the unique Line Group ID for the line.
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 Standard

Field Description
Incoming Default = Blank (Match any unspecified)
Number
Matches to the digits presented by the line provider. A blank record matches all calls that
do not match other records. By default this is a right-to-left matching. The options are:
• * = Incoming CLI Matching Takes Precedence
• – = Left-to-Right Exact Length Matching Using a - in front of the number causes a
left-to-right match. When left-to-right matching is used, the number match must be the
same length. For example -96XXX will match a DID of 96000 but not 9600 or 960000.
• X = Single Digit Wildcard Use X's to enter a single digit wild card character. For
example 91XXXXXXXX will only match DID numbers of at least 10 digits and starting
with 91, -91XXXXXXXX would only match numbers of exactly 10 digits starting with 91.
Other wildcard such as N, n and ? cannot be used.
Where the incoming number potentially matches two incoming call routes with X
wildcards and the number of incoming number digits is shorter than the number of
wildcards, the one with the shorter overall Incoming Number specified for matching is
used.
• i = ISDN Calling Party Number 'National' The i character does not affect the incoming
number matching. It is used for Outgoing Caller ID Matching, see notes below.
Incoming CLI Default = Blank (Match all)
Enter a number to match the caller's number (ICLID) provided with the call. This field is
matched left-to-right. The number options are:
• Full telephone number.
• Partial telephone number, for example just the area code.
• ! : Matches calls where the ICLID was withheld.
• ? : for number unavailable.
• For SIP call on a line using calling number verification, the characters P, F and Q can
be used to match calls that have passed authentication, failed authentication or were
unauthenticated respectively.
See SIP Calling Number Verification (STIR/SHAKEN) on page 870.
• Blank for all.

Call Setting Fields:

For calls routed using this Incoming Call Route, the settings of the following fields are applied to
the call regardless of the destination.
Field Description
Locale Default = Blank (Use system setting)
This option specifies the language prompts, if available, that voicemail should use for the call
if it is directed to voicemail.
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Incoming Call Route

Field Description
Priority Default = 1-Low. Range = 1-Low to 3-High.
This setting allows incoming calls to be assigned a priority. Other calls such as internal calls
are assigned priority 1-Low
In situations where calls are queued, high priority calls are placed before calls of a lower
priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where Voicemail Pro
is being used to provided queue ETA and queue position messages to callers since those
values will no longer be accurate when a higher priority call is placed into the queue. Note
also that Voicemail Pro will not allow a value already announced to an existing caller to
increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt group's
Queue Length Limit, the limit is temporarily raised by 1. This means that calls already
queued are not rerouted by the addition of a higher priority call into the queue.
A timer can be used to increase the priority of queued calls, see the setting System |
Telephony | Telephony | Call Priority Promotion Time.
The current priority of a call can be changed through the use of the p short code character in
a short code used to transfer the call.
Tag Default = Blank (No tag).
Allows a text tag to be associated with calls routed by this incoming call route. This tag is
displayed with the call within applications and on phone displays.
Hold Music Default = System source.
Source
The system can support several music on hold sources. See System | Telephony | Tones
and Music.
If the system has several hold music sources available, this field allows selection of the
source to associate with calls routed by this incoming call route. The new source selection
will then apply even if the call is forwarded or transferred away from the Incoming Call
Route destination. If the call is routed to another system in a multi-site network, the matching
source on that system is used if available. The hold music source associated with a call can
also be changed by a hunt group's Hold Music Source setting.
Ring Tone Default = Blank
Override
If ring tones have been configured in System | Telephony | Ring Tones, they are available
in this list. Setting a ring tone override applies a unique ring tone for the incoming call route.
Ring tone override features are only supported on 1400 Series, 9500 Series and J100 Series
(except J129) phones.

Outgoing Caller ID Matching

In cases where a particular Incoming Number is routed to a specific individual user, the system
will attempt to use that Incoming Number as the user's caller ID when they make outgoing calls if
no other number is specified. This requires that the Incoming Number is a full number suitable for
user as outgoing caller ID and acceptable to the line provider.
When this is the case, the character i can also be added to the Incoming Number field. This
character does not affect the incoming call routing. However when the same Incoming Number is

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 Voice Recording

used for an outgoing caller ID, the calling party number plan is set to ISDN and the type is set to
National. This option may be required by some network providers.
For internal calls being forwarded or twinned, if multiple incoming call route entries match the
extension number used as caller ID, the first entry created is used. This entry should start with
a “-” character (meaning fixed length) and provide the full national number. These entries do
not support wildcards. If additional entries are required for incoming call routing, they should be
created after the entry required for reverse lookup.
Related links
Incoming Call Route on page 521

Voice Recording
Navigation: Incoming Call Route | Voice Recording
These settings are used to activate the automatic recording of incoming calls that match the
incoming call route.
• Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
• Call recording starts when the call is answered.
• Call recording is paused when the call is parked or held. It restarts when the call is unparked
or taken off hold. This does not apply to SIP terminals.
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Recording continues for the duration of the call or up to the maximum recording time
configured on the voicemail server.
• Recording is stopped when the call ends or if:
- User call recording stops if the call is transferred to another user.
- Account code call recording stops if the call is transferred to another user.
- Hunt group call recording stops if the call is transferred to another user who is not a
member of the hunt group.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Record Inbound Default = None
Select whether automatic recording of incoming calls is enabled. The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. Otherwise, allow the call to continue without recording.
• Mandatory: Record the call if possible. Otherwise, block the call and return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
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Incoming Call Route

Field Description
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording is always active.
Recording Default = Mailbox
(Auto)
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This option set the destination for the recording to be a VRL
folder on the voicemail server. The VRL application polls that folder and collects waiting
recordings which it then places in its archive. Recording is still done by Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to the above but instructs
the voicemail server to create an authenticated recording. If the file contents are changed,
the file is invalidated though it can still be played.
- For systems recording to .opus format (the default), both settings create authenticated
recordings.

Related links
Incoming Call Route on page 521

Destinations
Navigation: Incoming Call Route | Destinations
The system allows multiple time profiles to be associated with an incoming call route. For each
time profile, a separate Destination and Fallback Extension can be specified.
When multiple records are added, they are resolved from the bottom up. The record used will be
the first one, working from the bottom of the list upwards, that is currently 'true', ie. the current
day and time or date and time match those specified by the Time Profile. If no match occurs the
Default Value options are used.
Once a match is found, the system does not use any other destination set even if the intended
Destination and Fallback Extension destinations are busy or not available.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.

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 Destinations

Field Description
Time Profile This column is used to specify the time profiles used by the incoming call routes. It displays
a drop-down list of existing time profiles from which a selection can be made. To remove an
existing entry, select it by clicking on the button on the left of the row, then right-click on the
row and select Delete.
The Default Value entry is fixed and is used if no match to a time profile below occurs.
Destination Default = Blank
Either enter the destination manually or select the destination for the call from the drop-down
list. The dr box which contains all available extensions, users, groups, RAS services and
voicemail. System short codes and dialing numbers can be entered manually. Once the
incoming call is matched the call is passed to that destination.
The following options appear in the drop-down list:
• Voicemail allows remote mailbox access with voicemail. Callers are asked to enter the
extension ID of the mailbox required and then the mailbox access code.
• Local user names.
• Local hunt groups names.
• AA: Name directs calls to an Embedded Voicemail auto-attendant services.
In addition to short codes, extension and external numbers, the following options can be also
be entered manually:
• VM:Name Directs calls to the matching start point in Voicemail Pro.
• A . matches the Incoming Number field. This can be used even when X wildcards are being
used in the Incoming Number field.
• A # matches all X wildcards in the Incoming Number field. For example, if the Incoming
Number was -91XXXXXXXXXXX, the Destination of # would match XXXXXXXXXXX.
• Text and number strings entered here are passed through to system short codes, for
example to direct calls into a conference. Note that not all short code features are
supported.
• If necessary, quote marks can be used to stop characters in the destination string being
interpreted as special characters.
Fallback Default = Blank (No fallback)
Extension
Defines an alternate destination which should be used when the current destination, set in
the Destination field cannot be obtained. For example if the primary destination is a hunt
group returning busy and without queuing or voicemail.

Related links
Incoming Call Route on page 521

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Chapter 35: WAN Port

These settings are used to configure the operation of system WAN ports and services.
WAN services can be run over a T1 PRI trunk connection. This requires creation of a virtual WAN
port. For full details refer to Using a Dedicated T1/PRI ISP Link.
Related links
WAN Port on page 530
Frame Relay on page 531
DLCIs on page 532
Advanced on page 533

WAN Port
Navigation: WAN Port | WAN Port
Use these settings to configure a WAN port.
On IP500 V2 systems, these settings configure the leased line connected to the WAN port on the
Control Unit. Normally this connection is automatically detected by the control unit. If a WAN Port
is not displayed, connect the WAN cable, reboot the Control Unit and receive the configuration.
The WAN Port configuration form is now be added.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Name The physical ID of the Extension port,. This parameter is not configurable; it is allocated
by the system.
Speed The operational speed of this port. For example for a 128K connection, enter 128000.
This should be set to the actual speed of the leased line as this value is used in the
calculation of bandwidth utilization. If set incorrectly, additional calls may be made to
increase Bandwidth erroneously.
Mode Default = SyncPPP
Select the protocol required. The options are:
• SyncPPP For a data link.
• SyncFrameRelay For a link supporting Frame Relay.
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 Frame Relay

Field Description
RAS Name If the Mode is SyncPPP, selects the RAS service to associate with the port. If the Mode
is SyncFrameRelay, the RAS Name is set through the DLCIs tab.

Related links
WAN Port on page 530

Frame Relay
Navigation: WAN Port | Frame Relay
These settings are for Frame Relay configuration.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Frame This must match the management type expected by the network provider. Selecting
Management Type AutoLearn allows the system to automatically determine the management type based
on the first few management frames received. If a fixed option is required the following
options are supported:
• Q933 AnnexA 0393
• Ansi AnnexD
• FRFLMI
• None
Frame Learn Mode This parameter allows the DLCIs that exist on the given WAN port to be provisioned in a
number of different ways.
• None No automatic learning of DLCIs. DLCIs must be entered and configured
manually.
• Mgmt Use LMI to learn what DLCIs are available on this WAN.
• Network Listen for DLCIs arriving at the network. This presumes that a network
provider will only send DLCIs that are configured for this particular WAN port.
• NetworkMgmt Do both management and network listening to perform DLCI learning
and creation.
Max Frame Length Maximum frame size that is allowed to traverse the frame relay network.
Fragmentation The options are:
Method
• RFC1490
• RFC1490+FRF12

Related links
WAN Port on page 530

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WAN Port

DLCIs
Navigation: WAN Port | DLCIs
DLCIs are created for Frame Relay connections. These settings are not mergeable. Changes to
these settings will require a reboot of the system.
Field Description
Frame Link Type Default = PPP
Data transfer encapsulation method. Set to the same value at both ends of the PVC
(Permanent Virtual Channel). The options are:
• None
• PPP Using PPP offers features such as out of sequence traffic reception, compression
and link level connection management.
• RFC 1490 RFC 1490 encapsulation offers performance and ease of configuration and
more inter-working with third party CPE.
• RFC1490 + FRF12 Alternate encapsulation to PPP for VoIP over Frame Relay. When
selected all parameters on the Service | PPP tab being used are overridden.
DLCI Default = 100 This is the Data Link Connection Identifier, a unique number assigned to
a PVC end point that has local significance only. Identifies a particular PVC endpoint
within a user's physical access channel in a frame relay.
RAS Name Select the RAS Service you wish to use.
Tc Default = 10
This is the Time Constant in milliseconds. This is used for measurement of data traffic
rates. The Tc used by the system can be shorter than that used by the network provider.
CIR (Committed Information Rate) Default = 64000 bps This is the Committed Information
Rate setting. It is the maximum data rate that the WAN network provider has agreed to
transfer. The committed burst size (Bc) can be calculated from the set Tc and CIR as Bc
= CIR x Tc. For links carrying VoIP traffic, the Bc should be sufficient to carry a full VoIP
packet including all its required headers. See the example below.
EIR (Excess Information Rate) Default = 0 bps This is the maximum amount of data in
excess of the CIR that a frame relay network may attempt to transfer during the given
time interval. This traffic is normally marked as De (discard eligible). Delivery of De
packets depends on the network provider and is not guaranteed and therefore they are
not suitable for UDP and VoIP traffic. The excess burst size (Be) can be calculated as
Be = EIR x Tc.

Example: Adjusting the Tc Setting

G.729 VoIP creates a 20 byte packet every 20ms. Adding typical WAN PPP headers results in a
33 byte packet every 20ms.
For a Committed Information Rate (CIR) of 14Kbps, with the Time Constant (Tc) set to 10ms; we
can calculate the Committed Burst size:
Bc = CIR x Tc = 14,000 x 0.01 = 140 bits = 17.5 bytes.

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 Advanced

Using 10ms as the Tc, a full G.729 VoIP packet (33 bytes) cannot be sent without exceeding the
Bc. The most likely result is lost packets and jitter.
If the Tc is increased to 20ms:
Bc = CIR x Tc = 14,000 x 0.02 = 280 bits = 35 bytes.
The Bc is now sufficient to carry a full G.729 VoIP packet.
Notes:
1. Backup over Frame Relay is not supported when the Frame Link Type is set to RFC1490.
2. When multiple DLCIs are configured, the WAN link LED is switched off if any of those
DLCIs is made inactive, regardless of the state of the other DLCIs. Note also that the WAN
link LED is switched on following a reboot even if one of the DLCIs is inactive. Therefore
when multiple DLCIs are used, the WAN link LED cannot be used to determine the current
state of all DLCIs.
3. When the Frame Link Type is set to RFC1490, the WAN link LED is switched on when
the WAN cable is attached regardless other whether being connected to a frame relay
network.
Related links
WAN Port on page 530

Advanced
Navigation: WAN Port | Advanced
These settings are used for Frame Relay connections.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Address Length The address length used by the frame relay network. The network provider will indicate if
lengths other than two bytes are to be used.
N391 Full Status Polling Counter
Polling cycles count used by the CPE and the network provider equipment when
bidirectional procedures are in operation. This is a count of the number of link integrity
verification polls (T391) that are performed (that is Status Inquiry messages) prior to a Full
Status Inquiry message being issued.
N392 Error Threshold Counter
Error counter used by both the CPE and network provider equipment. This value is
incremented for every LMI error that occurs on the given WAN interface. The DLCIs
attached to the given WAN interface are disabled if the number of LMI errors exceeds this
value when N393 events have occurred. If the given WAN interface is in an error condition
then that error condition is cleared when N392 consecutive clear events occur.
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WAN Port

Field Description
N393 Monitored Events Counter
Events counter measure used by both the CPE and network provider equipment. This
counter is used to count the total number of management events that have occurred in
order to measure error thresholds and clearing thresholds.
T391 Link Integrity Verification Polling Timer
The link integrity verification polling timer normally applies to the user equipment and
to the network equipment when bidirectional procedures are in operation. It is the time
between transmissions of Status Inquiry messages.
T392 Polling Verification Timer The polling verification timer only applies to the user equipment
when bidirectional procedures are in operation. It is the timeout value within which to
receive a Status Inquiry message from the network in response to transmitting a Status
message. If the timeout lapses an error is recorded (N392 incremented).

Related links
WAN Port on page 530

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Chapter 36: Directory Entry

Navigation: Directory | Directory Entry

Additional configuration information
For additional configuration information, see Centralized System Directory on page 612.
Configuration settings
Use these settings to create directory records that are stored in the system's configuration. Directory
records can also be manually imported from a CSV file. The system can also use Directory Services
to automatically import directory records from an LDAP server at regular intervals.
A system can also automatically import directory records from another system. Automatically
imported records are used as part of the system directory but are not part of the editable
configuration. Automatically imported records cannot override manually entered records.
For a Server Edition network, these settings can only be configured at the network level and they
are stored in the configuration of the Primary Server. All other systems in the network are configured
to share the directory settings of the Primary Server through their Manager settings at System |
Directory Services | HTTP.
Directory Special Characters
The following characters are supported in directory records. They are supported in both system
configuration records and in imported records.
• ? = Any Digit Directory records containing a ? are only used for name matching against the
dialed or received digits on outgoing or incoming calls. They are excluded from the dialable
directory. In the following example, any calls where the dialed or received number that starts
9732555 will have the display name Homdel associated with them.
- Name: Holmdel
- Number: 9732555?
• ( ) = Optional Digits Brackets can be used to enclose an optional portion of a number, typically
the area code. Only one pair of brackets are supported in a number. Records containing digits
inside ( ) brackets are only used for user dialling. The full string is dialed with the ( ) brackets
removed.
• - Characters Directory records can also contain - characters. Records containing - characters
are only used for user dialing. The full string is dialed with the - character removed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Directory Entry

Field Description
Index Range = 000 to 999 or None.
This value is used with system speed dials dialed from M and T-Series phones. The value
can be changed but each value can only be applied to one directory record at any time.
Setting the value to None makes the speed dial inaccessible from M and T-Series phones,
however it may still be accessible from the directory functions of other phone types and
applications. The Speed Dial short code feature can be used to create short codes to dial
the number stored with a specific index value.
Name Enter the text, to be used to identify the number. Names should not begin with numbers.
Number Enter the number to be matched with the above name. The number is processed against
the applicable user and system short codes. Note that if the system has been configured
to use an external dialing prefix, that prefix should be added to directory numbers.

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Chapter 37: Time Profile

Navigation: Time Profile | Time Profile

Additional configuration information
This section provides the Time Profiles field descriptions. For additional configuration information,
see:
• Configuring Time Profiles on page 667
• The button action Time Profile on page 1146
Configuration settings
For a time profile with multiple records, for example a week pattern and some calendar records, the
profile is valid when any entry is valid. For Server Edition, this type of configuration record can be
saved as a template and new records created from a template.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
• For systems using record consolidation, you can only add and edit this type of record at the
solution level. The record is then automatically copied to each IP Office system in the network.
Field Description
Name Range = Up to 15 characters
This name is used to select the time profile from within other tabs.
Manual Override Default = Off.
You can manually override a time profile. The override settings allow you to mix timed and
manual settings. The options are:
• Active Until Next Timed Inactive: Use for time profiles with multiple intervals. Select to
make the current timed interval active until the next inactive interval.
• Inactive Until Next Timed Active: Use for time profiles with multiple intervals. Select to
make the current active timed interval inactive until the next active interval.
• Latch Active: Set the time profile to active. Timed inactive periods are overridden and
remain active. The setting is retained over a reboot.
• Latch Inactive: Set the time profile to inactive. Timed active periods are overridden and
remain active. The setting is retained over a reboot.
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Time Profile

Field Description
Time Entry List
This list shows the current periods during which the time profile is active. Clicking on an existing entry will
display the existing settings and allows them to be edited if required. To remove an entry, selecting it and then
click on Remove or right-click and select Delete.
Recurrence When a new time entry is required, click Add Recurring and then enter the settings
Pattern (Weekly for the entry using the fields displayed. Alternately right-click and select Add Recurring
Time Pattern) Time Entry. This type of entry specific a time period and the days on which it occurs, for
example 9:00 - 12:00, Monday to Friday. A time entry cannot span over two days. For
example you cannot have a time profile starting at 18:00 and ending 8:00. If this time
period is required two Time Entries should be created - one starting at 18:00 and ending
11:59, the other starting at 00:00 and ending 8:00.
• Start Time The time at which the time period starts.
• End Time The time at which the time period ends. Note that the endtime is at the end of
the minute, for example 11:00 is interpreted as 11:00:59, not 11:00:00.
• Days of Week The days of the week to which the time period applies.
Recurrence When a new calendar date entry is required, click Add Date and then enter the settings
Pattern (Calendar required. Alternately right-click and select Add Calendar Time Entry. Calendar records
Date) can be set for up to the end of the next calendar year.
• Start Time The time at which the time period starts.
• End Time The time at which the time period ends.
• Year Select either the current year or the next calendar year.
• Date To select or de-select a particular day, double-click on the date. Selected days are
shown with a dark gray background. Click and drag the cursor to select or de-select a
range of days.

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Chapter 38: Firewall Profile

The IP Office system can act as a firewall, allowing only specified data traffic to start a session
across the firewall and controlling in which direction such sessions can be started.
You can select a firewall profiles for the following areas of IP Office operation:
• You can apply a firewall profile to traffic between LAN1 and LAN2.
• You can select a firewall for users who are the destination of incoming RAS calls.
• You can select a firewall when you configure a service.
Note:
• The IP Office firewall profiles can include Static network address translation (NAT) records.
If the firewall profile contains any Static NAT records, the IP Office blocks traffic that does
not match one of those static NAT records.
• If Network Address Translation (NAT) is used with the firewall, you must configure the
Primary Trans. IP Address setting on incoming services (Service | IP | Primary Trans. IP
Address ).
• On Linux-based systems, to ensure that the firewall starts after a reboot, you must
enable the Solution > > Platform View > Settings > System > Firewall Settings >
Activateoption.
Related links
Firewall | Standard on page 539
Firewall | Custom on page 541
Static NAT on page 543

Firewall | Standard
Navigation: Firewall Profile | Standard
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Configuration settings
By default, any protocol not listed in the standard firewall list is dropped unless a custom firewall
entry is configured for that protocol.

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Firewall Profile

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Name Range = Up to 15 characters. Enter the name to identify this profile.
Protocol For each of the listed protocols, the options Drop, In (Incoming traffic can start a session),
Control Out (Outgoing traffic can start a session) and Both Directions can be selected. Once a
session is started, return traffic for that session is also able to cross the firewall.
Protocol Default Description
TELNET Out Remote terminal log in.
FTP Out File Transfer Protocol.
SMTP Out Simple Mail Transfer Protocol.
TIME Out Time update protocol.
DNS Out Domain Name System.
GOPHER Drop Internet menu system.
FINGER Drop Remote user information protocol.
RSVP Drop Resource Reservation Protocol.
HTTP/S Bothway Hypertext Transfer Protocol.
POP3 Out Post Office Protocol.
NNTP Out Network News Transfer Protocol.
SNMP Drop Simple Network Management Protocol.
IRC Out Internet Relay Chat.
PPTP Drop Point to Point Tunneling Protocol.
IGMP Drop Internet Group Membership Protocol.
Service Control For each of the listed services, the options Drop, In, Out and Both Directions can be
selected. Once a session is started, return traffic for that session is also able to cross the
firewall.
Protocol Default Description
SSI In System Status Application access.
SEC Drop TCP security settings access.
CFG Drop TCP configuration settings access.
TSPI In TSPI service access.
WS Drop IP Office web management services.

Related links
Firewall Profile on page 539

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 Firewall | Custom

Firewall | Custom
Navigation: Firewall Profile | Custom
The tab lists custom firewall settings added to the firewall profile. The Add, Edit and Remove
controls can be used to amend the settings in the list.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Notes For information only. Enter text to remind you of the purpose of the custom firewall record.
Remote IP The IP address of the system at the far end of the link. Blank allows all IP addresses.
Address
Remote IP Mask The mask to use when checking the Remote IP Address. When left blank no mask is set,
equivalent to 255.255.255.255 - allow all.
Local IP Address The address of devices local to this network (pre-translated). Blank allows all IP addresses.
Local IP Mask The mask to use when checking the Local IP Address. When left blank no mask is set,
equivalent to 255.255.255.255 - allow all.
IP Protocol The value entered here corresponds to the IP Protocol which is to be processed by this
Firewall profile: 1 for ICMP, 6 for TCP, 17 for UDP or 47 for GRE. This information can be
obtained from the "pcol" parameter in a Monitor trace.
Match Offset The offset into the packet (0 = first byte of IP packet) where checking commences for either
a specific port number, a range of port numbers, or data.
Match Length The number of bytes to check in the packet, from the Match Offset point, that are checked
against the Match Data and Match Mask settings.
Match Data The values the data must equal once masked with the Match Mask. This information can
be obtained from "TCP Dst" parameter in a Monitor trace (the firewall uses hex so a port
number of 80 is 50 in hex)
Match Mask This is the byte pattern, which is logically ANDed with the data in the packet from the offset
point. The result of this process is then compared against the contents of the "Match Data"
field.
Direction The direction that data may take if matching this filter.
Drop All matching traffic is dropped.
In Incoming traffic can start a session.
Out Outgoing traffic can start a session.
Both Directions Both incoming and outgoing traffic can start sessions.

Example Custom Firewall Records

Dropping NetBIOS searches on an ISPs DNS:
We suggest that the following filter is always added to the firewall facing the Internet to avoid
costly but otherwise typically pointless requests from Windows machines making DNS searches
on the DNS server at your ISP.
• Direction: Drop
• IP Protocol: 6 (TCP)

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Firewall Profile

• Match Offset: 20
• Match Length: 4
• Match Data: 00890035
• Match Mask: FFFFFFFF
Browsing Non-Standard Port Numbers
The radio button for HTTP permits ports 80 and 443 through the firewall. Some hosts use non-
standard ports for HTTP traffic, for example 8080, 8000, 8001, 8002, etc. You can add individual
filters for these ports as you find them.
You wish to access a web page but you cannot because it uses TCP port 8000 instead of the
more usual port 80, use the entry below.
• Direction: Out
• IP Protocol: 6 (TCP)
• Match Offset: 22
• Match Length: 2
• Match Data: 1F40
• Match Mask: FFFF
A more general additional entry given below allows all TCP ports out.
• Direction: Out
• IP Protocol: 6 (TCP)
• Match Offset: 0
• Match Length: 0
• Match Data: 00000000000000000000000000000000
• Match Mask: 00000000000000000000000000000000
Routing All Internet Traffic through a WinProxy
If you wish to put WinProxy in front of all Internet traffic via the Control Unit. The following firewall
allows only the WinProxy server to contact the Internet : -
1. Create a new Firewall profile and select Drop for all protocols
2. Under Custom create a new Firewall Entry
3. In Notes enter the name of the server allowed. Then use the default settings except in
Local IP Address enter the IP address of the WinProxy Server, in Local IP Mask enter
255.255.255.255 and in Direction select Both Directions.
Stopping PINGs
You wish to stop pings - this is ICMP Filtering. Using the data below can create a firewall filter that
performs the following; Trap Pings; Trap Ping Replies; Trap Both.
• Trap Pings: Protocol = 1, offset = 20, data = 08, mask = FF
• Trap Ping Replies: Protocol = 1, offset = 20, data = 00, mask = FF

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 Static NAT

• Trap Both: Protocol = 1, offset = 20, data = 00, mask = F7, Traps Both.
Related links
Firewall Profile on page 539

Static NAT
Navigation: Firewall Profile | Static NAT
The Static NAT table allows the firewall to perform address translation between selected internal
and external IP addresses. Up to 64 internal and external IP address pairs can be added to the
Static NAT section of a Firewall Profile.
This feature is intended for incoming maintenance access using applications such as PC-
Anywhere, Manager and the Voicemail Pro Client. The address translation is used for destinations
such a Voicemail Pro server or the system's own LAN1 address.
• If there are any records in the Static NAT settings of a Firewall Profile, each packet
attempting to pass through the firewall must match one of the static NAT pairs or else the
packet will be dropped.
• The destination address of incoming packets is checked for a matching External IP
Address. If a match is found, the target destination address is changed to the corresponding
Internal IP Address.
• The source address of outgoing packets is checked for a matching Internal IP Address. If a
match is found, the source address is changed to the corresponding External IP Address.
• Even when a static NAT address match occurs, the other settings on the Firewall Profile
Standard and Custom tabs are still applied and may block the packet.
Related links
Firewall Profile on page 539

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Chapter 39: IP Route

Additional configuration information

This section provides the IP Route field descriptions. For additional configuration information, see
Configuring IP Routes on page 622.
Related links
IP Route | IP Route on page 544
RIP Dynamic Routing on page 545

IP Route | IP Route
Navigation: IP Route | IP Route
Additional configuration information
For additional configuration information, see Configuring IP Routes on page 622.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 686.
Configuration settings
These settings are used to setup static IP routes from the system. These are in addition to RIP if
RIP is enabled on LAN1 and or LAN2. Up to 100 routes are supported.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
Warning:
• The process of 'on-boarding' (refer to the Deploying Avaya IP Office™ Platform SSL VPN
Services manual) may automatically add a static route to an SSL VPN service in the
system configuration when the on-boarding file is uploaded to the system. Care should
be taken not to delete or amend such a route except when advised to by Avaya.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
IP Address The IP address to match for ongoing routing. Any packets meeting the IP Address and IP
Mask settings are routed to the entry configured in the Destination field. When left blank
then an IP Address of 255.255.255.255 (all) is used.
Table continues…

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 RIP Dynamic Routing

Field Description
IP Mask The subnet mask used to mask the IP Address for ongoing route matching. If blank, the
mask used is 255.255.255.255 (all).
A 0.0.0.0 entry in the IP Address and IP Mask fields routes all packets for which there is
no other specific IP Route available. The Default Route option with Services can be used
to do this if a blank IP route is not added.
Gateway IP Default = Blank The address of the gateway where packets for the above address are to
Address be sent. If this field is set to 0.0.0.0 or is left blank then all packets are just sent down
to the Destination specified, not to a specific IP Address. This is normally only used to
forward packets to another Router on the local LAN.
Destination Allows selection of LAN1, LAN2 and any configured Service, Logical LAN or Tunnel
(L2TP only).
Metric: Default = 0
The number of "hops" this route counts as.
Proxy ARP Default = Off
This allows the system to respond on behalf of this IP address when receiving an ARP
request.

Related links
IP Route on page 544

RIP Dynamic Routing

Navigation: IP Route | RIP Dynamic Routing
Routing Information Protocol (RIP) is a protocol which allows routers within a network to exchange
routes of which they are aware approximately every 30 seconds. Through this process, each
router adds devices and routes in the network to its routing table.
Each router to router link is called a 'hop' and routes of up to 15 hops are created in the routing
tables. When more than one route to a destination exists, the route with the lowest metric (number
of hops) is added to the routing table.
When an existing route becomes unavailable, after 5 minutes it is marked as requiring 'infinite' (16
hops). It is then advertised as such to other routers for the next few updates before being removed
from the routing table. The system also uses 'split horizon' and 'poison reverse'.
RIP is a simple method for automatic route sharing and updating within small homogeneous
networks. It allows alternate routes to be advertised when an existing route fails. Within a large
network the exchange of routing information every 30 seconds can create excessive traffic. In
addition the routing table held by each system is limited to 100 routes (including static and internal
routes).

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IP Route

It can be enabled on LAN1, LAN2 and individual services. The normal default is for RIP to be
disabled.
• Listen Only (Passive): The system listens to RIP1 and RIP2 messages and uses these to
update its routing table. However the system does not respond.
• RIP1: The system listens to RIP1 and RIP2 messages. It advertises its own routes in a RIP1
sub-network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): The system listens to RIP1 and RIP2 messages. It
advertises its own routes in a RIP2 sub-network broadcast. This method is compatible with
RIP1 routers.
• RIP2 Multicast: The system listens to RIP1 and RIP2 messages. It advertises its own routes
to the RIP2 multicast address (249.0.0.0). This method is not compatible with RIP1 routers.
Broadcast and multicast routes (those with addresses such as 255.255.255.255 and 224.0.0.0)
are not included in RIP broadcasts. Static routes (those in the IP Route table) take precedence
over a RIP route when the two routes have the same metric.
Related links
IP Route on page 544

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Chapter 40: Account Code

Additional configuration information

This section provides the Account Code field descriptions. For additional configuration information,
see Configuring Account Codes on page 722.
Account codes are commonly used to control cost allocation and out-going call restriction. The IP
Office can use account codes in a number of ways.
• When making calls, users can voluntarily enter an account code.
- On phones that support programmable buttons, users can do this using an Account Code
Entry button.
- A short code set to Set Account Code can also be used to enter an account code before
making a call.
• If the number dialed for an outgoing call matches a short code set to Forced Account Code,
the user is required to enter a valid account code in order to continue the call.
• Individual users can be set to Forced Account Code (User > Telephony > Supervisor
Settings). They then need to enter an account code for any outgoing external calls.
• Incoming calls can also be associated with an account code by matching the Caller ID stored
with the account code settings. That account code is then included in the call's SMDR call log.
When an account code is entered during a call:
• The IP Office checks the code entered for a match against those account codes set in its
configuration. For Forced Account Code calls, the call is not allowed until a valid code is
entered.
• If the code is valid, it is included in the information output by the system's SMDR call log.
• The account code used on a call is not included in the user's personal call log. This means that
re-dial functions will not re-enter the account code.
• If more than one account code is entered during a call, only the last code entered is included in
the SMDR call log.
An IP Office system can support up to 1500 configured account codes.
• Wildcards can be used in the account codes configured to expand the supported range. For
example, a single account code entry 9?? allows dialing any number between 900 and 999 to
be treated as a valid account code.
• By default, in Server Edition/Select networks, account codes are configured at the network
level and automatically replicated in the configuration of all systems in the network. That is,
the 1500 account code limit applies to the whole network. They can only be seen and edited
at the individual system configuration level if record consolidation is switched off. See Record
Consolidation on page 70.

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Account Code

Related links
Account Code on page 548
Voice Recording on page 548

Account Code
Navigation: Account Code | Account Code
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Descriptions
Account Code Enter the account code required. It can also include wildcards; ? matches a single digit
and * matches any digits.
Caller ID A caller ID can be entered and used to automatically assign an account code to calls
made to or received from caller ID.

Related links
Account Code on page 547

Voice Recording
Navigation: Account Code | Voice Recording
These settings are used to activate the automatic recording of external calls when the account
code is entered at the start of the call.
• Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
• Call recording starts when the call is answered.
• Call recording is paused when the call is parked or held. It restarts when the call is unparked
or taken off hold. This does not apply to SIP terminals.
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Recording continues for the duration of the call or up to the maximum recording time
configured on the voicemail server.
• Recording is stopped when the call ends or if:
- User call recording stops if the call is transferred to another user.
- Account code call recording stops if the call is transferred to another user.

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 Voice Recording

- Hunt group call recording stops if the call is transferred to another user who is not a
member of the hunt group.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Record Outbound Default = None
Select whether automatic recording of outgoing calls is enabled. The Auto Record Calls
option sets whether just external calls or external and internal calls are included. The
options are:
• None: Do not automatically record calls.
• On: Record the call if possible. Otherwise, allow the call to continue without recording.
• Mandatory: Record the call if possible. Otherwise, block the call and return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording is always active.
Recording (Auto) Default = Mailbox
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This option set the destination for the recording to be a VRL
folder on the voicemail server. The VRL application polls that folder and collects waiting
recordings which it then places in its archive. Recording is still done by Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to the above but
instructs the voicemail server to create an authenticated recording. If the file contents
are changed, the file is invalidated though it can still be played.
- For systems recording to .opus format (the default), both settings create
authenticated recordings.

Related links
Account Code on page 547

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Chapter 41: Subscription

Navigation: Subscription| Subscription

Subscriptions are monthly paid entitlements used by subscription mode systems. They can be
divided into two main groups; per-user per-month user subscriptions and per-month application
subscriptions. For more information, see Subscriptions on page 604.
Subscription are ordered from the Avaya Channel Marketplace, using the unique ID number of the
system. Once ordered, details of the customer number and address of the subscription server are
supplied in an email. Those details are then used during the initial system configuration.
Field Descriptions
System ID The unique number used for validation of subscriptions.
• For IP500 V2 systems, this is the System SD card’s PLDS ID as printed on
the card. For older cards with 10-digits ID, the number is prefixed with 11.
• For other systems, the ID is a unique value based on elements of the system
hardware at the time of system installation.
Customer ID This number is supplied in the email provided when the System ID is
subscribed with Avaya.
Customer Name The customer name used when the System ID was subscribed.
License Server This address is supplied in the email provided when the System ID is
subscribed with Avaya.

Available Subscriptions
The following subscriptions can be ordered for an IP Office Subscription system.
Table 2: User Subscriptions

Subscription Description
Telephony User Enables a user with telephony functions using a deskphone.
Telephony Plus User Enable a user with telephony functions using an deskphone and or a softphone
client on a PC.
UC User Enable a user with the full range of telephony functions.

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Table 3: Application Subscriptions

Subscription Description
Receptionist Console Enables use of the IP Office SoftConsole application to answer and redirect
calls. The number of subscriptions allows the matching number of users
to be configured as Receptionist users. Those users still require a user
subscriptions for their telephone connection (IP Office SoftConsole is not a
softphone).
Media Manager This subscription enables support for Media Manager. This uses Voicemail
Pro to perform call recording. Media Manager then collects and stores those
recordings. Media Manager can be provided as a local or centralized service as
follows:
• Run locally on the same server as the Voicemail Pro service and storing the
recordings on an additional hard disk installed in that server.
• Run centralized and storing the recordings on the cloud-based servers
providing the system's subscriptions. In this case, the number of
subscriptions also controls the maximum number of recordings supported:
1. 150,000
2. 300,000
3. 500,000
4. 750,000
5. 1,000,000
Third-Party CTI This subscription enables support for CTI connections by third-party
applications. This includes DevLink, DevLink3, 3rd-party TAPI and TAPI WAV.
Avaya Contact Center This subscription enables support the Avaya Contact Center Select (ACCS)
Select service hosted on a separate server.
Avaya Call Reporter This subscription enables support for the Avaya Call Reporter application,
hosted on a separate server.

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Chapter 42: License

Additional configuration information

This section provides the Licenses field descriptions.
Note:
This section is not applicable to systems running in subscription mode.
For additional configuration information on licensing, see the following.
• Applying Licenses on page 670.
• Converting from Nodal Licensing to Centralized Licensing on page 682
• Migrating ADI Licenses to PLDS on page 684
• “Licenses” in Avaya IP Office™ Platform Solution Description.
Related links
License on page 552
Remote Server on page 555

License
Navigation: License | License
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 License

Name Description
License Mode Identifies the status of the system licenses. The two license configuration types are nodal
and WebLM. Nodal licenses are licenses that are present on the system. WebLM licenses
means licenses obtained from the WebLM server.

The possible states are:

• Normal Mode Normal nodal licensing mode. In this mode, WebLM is not configured and
only nodal licensing is allowed.
• Server Error This mode occurs when transitioning to WebLM licensing. WebLM has
been configured but the server is not available.
• Configuration Error This mode occurs when transitioning to WebLM licensing. WebLM
has been configured and the server is available, but there are not enough licenses
available to license all of the configured features. Only nodal licenses are valid on
Standard mode IP500 V2 systems.
• WebLM Normal Mode The system is fully licensed. WebLM has been configured and
there are enough licenses available to license all of the configured features.
• WebLM Error Mode Action is required to correct the License Mode. Refer to the License
Status column and the Error List section at the bottom of the screen to determine why
the system is in License Error Mode. A 30-day grace period provides access to the
capacities and features of the installed license when the system is in License Error Mode.
• WebLM Restricted Mode When the system is in License Error Mode, if the problem is
not resolved with the 30-day grace period, the system will enter License Restricted Mode.
When in this mode, configuration changes are blocked, except for fixing the licensing
errors. If a feature license cannot be acquired from the WebLM server, the feature will not
function.
Type Mode WebLM Configured Virtual License and
Grace Period (30
days)
Nodal Normal
WebLM Server Error
WebLM Configuration Error
WebLM Normal
WebLM Error
WebLM Restricted
Licensed Version Indicates the software version the system is currently licensed for.
PLDS Host ID The ID used when generating PLDS nodal license files.
Not used with WebLM licensing. WebLM licensing uses the host ID of the WebLM server.
PLDS File Status If a PLDS nodal license file is loaded, this field indicates if the file is valid or not.
Select Licensing Indicates that the system has a valid Select license.
Feature Identifies the licenses installed on the system.
Table continues…

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License

Name Description
Key This is the license key string supplied. It is a unique value based on the feature being
licensed and the either the system's Dongle Serial Number or System Identification
depending on the type of system.
Not applicable when using PLDS or WebLM licensing. This field is not displayed if there are
no ADI licenses.
Instance For information only. Some licenses enable a number of port, channels or users. When that
is the case, the number of such is indicated here. Multiple licenses for the same feature are
usually cumulative.
Status For information only. This field indicates the current validation status of the license key.
• Unknown This status is shown for licenses that have just been added to the
configuration shown in Manager. Once the configuration has been sent back to the
system and then reloaded, the status will change to one of those below.
• Valid: The license is valid.
• Invalid: The license was not recognized. It did not match the PLDS host ID.
• Dormant: The license is valid but is conditional on some other pre-requisite licenses.
• Obsolete: The license is valid but is one no longer used by the level of software running
on the system.
Expiry Date For information only. Trial licenses can be set to expire within a set period from their issue.
The expiry date is shown here.
Source The source of the license file. The options are:
• ADI Nodal: ADI licenses added locally to the system. This may appear on upgraded
systems.
• PLDS Nodal: PLDS licenses added locally to the system.
• WebLM: Licenses obtained from the WebLM server.
• Virtual: Licenses created by the system. This may appear on upgraded systems.
• Virtual Grace: Licenses created by the system while in WebLM error mode.

Additional Configuration Information

Click PLDS License > Send To IP Office > OK to open the Select PLDS License File dialog
from where you can upload a PLDS license to IP Office. You can browse to a location on your
system and select a file to upload.
Select an existing license and click PLDS License > Delete From IP Office > OK to delete the
selected license.
Related links
License on page 552

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 Remote Server

Remote Server
Navigation: License | Remote Server
This tab is used for:
• For licensing of IP500 V2 systems in Enterprise Branch deployments.
• To specify the method of centralized licensing used for PLDS licensed IP Office Server
Edition and Select systems
The Reserved Licenses setting is mergeable. The remaining settings are not mergeable.
Changes to these settings requires a reboot of the system.
The following fields control which source the IP Office system uses for its licenses. The fields
shown depends on the type of IP Office system:
Field Description
Licence Source Default = WebLM.
This field is available on Server Edition systems. All IP Office systems in the network
must use the same source for licensing. The options are:
• WebLM
Use licenses obtained from the PLDS license file uploaded to the WebLM service. All
IP Office servers in the network request license reservations from the WebLM service.
- On Server Edition systems, a Deploy button appears when you select WebLM as
License Source. Click the Deploy button to browse and select a license file to
deploy.
• Local / Primary Server
PLDS license files are uploaded to the IP Office service of each IP Office server.
Depending on the particular licenses:
- Some licenses are obtained by reservation requests to the primary IP Office server
- Others licenses are obtained from the IP Office server’s own license file.
Enable Remote Default = Off.
Server
This field is available on non-Server Edition IP500 V2 systems. The options are:
• If disabled, the IP Office system is licensed locally by uploading a license file to the
system.
• If enabled, the system uses licenses requested from a remote WebLM server. This
option is only supported for systems in a branch enterprise supported via Avaya Aura®
System Manager.

The additional fields displayed depend on the license source selection above:

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License

Local/Primary Server Licensed Server Settings

Field Description
License Server IP Default = 127.0.0.1 on Primary. On Secondary and expansion systems, the default is the
Address Primary IP address.
This field is available when the Licence Source is set to Local Primary Server. This
field contains the IP address of the Server Edition Primary server.

WebLM Licensed Primary Server Settings

Field Description
Domain Name Default = Blank for IP500 V2 systems and for Server Edition Primary hosted
(URL) deployments. For Server Edition, the IP address of the Primary Server.
• For Enterprise Branch deployments, the domain name or IP address of the WebLM
server or the domain name of System Manager if the system is under System Manager
control.
• For Server Edition deployments, the domain name or IP address of the Primary Server.
• For Server Edition hosted deployments, the domain name of the WebLM server.

The format can be the FQDN or the IP address prefixed with https://.
Path Default = WebLM/LicenseServer.
The path on the web server of the WebLM resource.
Port Number Default = 52233.
The port number of the WebLM server.
WebLM Client ID Default = An ID based on MAC address and hostname of the system.
This field is editable but must only be changed on Avaya Partner Hosted systems.

Important:
• For all other deployment scenarios, you must ensure the value matches the
WebLM Node ID below. Changing the value will cause license issues for users
and phones, including non-IP phones.
WebLM Node ID Default = An ID based on MAC address and hostname of the system. Read-only.
Used by the WebLM service to identify the IP Office system requesting licenses.

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 Remote Server

WebLM Licensed Server (non-Primary) Settings

Field Description
Enable proxy via Default = On.
Primary IP Office
Available on Server Edition Secondary and Expansion systems.
line
• Enables retrieval of licenses from the WebLM server through the IP Office Line
connection to the Server Edition Primary server.
• If the check box is cleared, the WebLM request is done directly to the WebLM server.
Note that this field is not available if the node is not configured as a WebSocket client to
the Server Edition Primary server.
Primary IP Address Default = The IP address of the Server Edition Primary server.
Available on Server Edition Secondary and Expansion systems when Enable proxy via
Primary IP Office line is enabled
WebLM Client ID Default = An ID based on MAC address and hostname of the system. Read-only.
WebLM Node ID Used by the WebLM service to identify the IP Office system requesting licenses.

Reserved Licenses
These fields are used to reserve licenses from the license server, WebLM or, if using nodal
licensing, the Primary server. There are two types of reservation field; manual and automatic.
• Manual fields can be used to set the number of licenses that the server should request from
those available on the primary/WebLM server.
• Automatic fields are set to match other aspects of the server configuration, for example the
number of configured power users. Note that these values may not change until after the
configuration is saved and then reloaded.
WebLM Reserved Licenses — Primary Server Secondary Expansion Expansion
Manual Server (Linux) (IP500 V2)
SIP Trunk Sessions
SM Trunk Sessions
Voicemail Pro Ports - -
VMPro Recordings - -
Administrators
VMPro TTS Professional - -
Wave Users - - -
CTI Link Pro
UMS Web Services
MAC Softphones
Avaya Contact Center Select - -
Third Party Recorder - -
VM Media Manager -
Table continues…

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License

Customer Service Supervisor

Customer Service Agent

Nodal Reserved Licenses — Primary Server Secondary Expansion Expansion

Manual Server (Linux) (IP500 V2)
SIP Trunk Sessions

WebLM/Nodal Reserved Licenses Primary Server Secondary Expansion Expansion

— Automatic Server (Linux) (IP500 V2)
Server Edition
Avaya IP Endpoints
3rd-Party IP Endpoints
Receptionist
Office Worker
Power User
Avaya Softphone
Web Collaboration
Universal PRI Additional - - -
Channels
IPSec Tunneling - -

Related links
License on page 552

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Chapter 43: Tunnel

Tunneling allows additional security to be applied to IP data traffic. This is useful when sites across
an unsecure network such as the public internet. The IP500 V2 system supports two methods of
tunneling, L2TP and IPSec. Once a tunnel is created, it can be used as the destination for selected
IP traffic in the IP Route table.
• The use of tunnels is only supported on non-Subscription IP Office IP500 V2 systems.
Type Description
L2TP Layer 2 Tunneling Protocol PPP (Point to Point Protocol) authentication normally takes
place between directly connected routing devices. For example when connecting to the
internet, authentication is between the customer router and the internet service provider's
equipment. L2TP allows additional authentication to be performed between the routers at
each end of the connection regardless of any intermediate network routers. The use of
L2TP does not require a license.
IPSec IPSec allows data between two locations to be secured using various methods of sender
authentication and or data encryption. The use of IPSec requires entry of an IPSec
Tunneling license into the system at each end.

Related links
L2TP Tunnel on page 559
IP Security Tunnel on page 562

L2TP Tunnel
Layer 2 Tunneling Protocol PPP (Point to Point Protocol) authentication normally takes place
between directly connected routing devices. For example when connecting to the internet,
authentication is between the customer router and the internet service provider's equipment.
L2TP allows additional authentication to be performed between the routers at each end of the
connection regardless of any intermediate network routers. The use of L2TP does not require a
license.

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Tunnel

Related links
Tunnel on page 559
L2PT Tunnel on page 560
L2TP on page 561
L2TP PPP on page 561

L2PT Tunnel
Navigation: Tunnel | Tunnel (L2TP)
Configuration settings
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Name Default = Blank.
A unique name for the tunnel. Once the tunnel is created, the name can be selected as a
destination in the IP Route table.
Local Configuration
The account name and password is used to set the PPP authentication parameters.
Local Account The local user name used in outgoing authentication.
Name
Local Account The local user password. Used during authentication.
Password/
Confirm
Password
Local IP Address The source IP address to use when originating an L2TP tunnel. By default (un-configured),
the system uses the IP address of the interface on which the tunnel is to be established as
the source address of tunnel.

Remote Configuration
The account name and password is used to set the PPP authentication parameters.
Remote Account The remote user name that is expected for the authentication of the peer.
Name
Remote Account The password for the remote user. Used during authentication.
Password/
Confirm
Password
Remote IP The IP address of the remote L2TP peer or the local VPN line IP address or the WAN IP
Address address.
Minimum Call Default = 60 minutes. Range = 1 to 999.
Time (Mins)
The minimum time that the tunnel will remain active.
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 L2TP Tunnel

Field Description
Forward Default = On
Multicast
Allow the tunnel to carry multicast messages when enabled.
Messages
Encrypted Default = Off
Password
When enabled, the CHAP protocol is used to authenticate the incoming peer.

Related links
L2TP Tunnel on page 559

L2TP
Navigation: Tunnel | L2TP
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Shared Secret/ User setting used for authentication. Must be matched at both ends of the tunnel. This
Confirm password is separate from the PPP authentication parameters defined on the L2TP|Tunnel
Password tab.
Total Control Default = 0. Range = 0 to 65535.
Retransmission
Time delay before retransmission.
Interval
Receive Window Default = 4. Range = 0 to 65535.
Size
The number of unacknowledged packets allowed.
Sequence Default = On
numbers on Data
When on, adds sequence numbers to L2TP packets.
Channel
Add checksum Default = On.
on UDP packets
When on, uses checksums to verify L2TP packets.
Use Hiding Default = Off
When on, encrypts the tunnel's control channel.

Related links
L2TP Tunnel on page 559

L2TP PPP
Navigation: Tunnel | PPP (L2TP)
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Tunnel

Field Description
CHAP Challenge Default = 0 (Disabled). Range = 0 to 99999 seconds.
Interval (secs)
Sets the period between CHAP challenges. Blank or 0 disables repeated challenges.
Header Default = None
Compression
Select header compression. Options are: IPHC and/or VJ.
PPP Compression Default = MPPC
Mode
Select the compression mode for the tunnel connection. Options are: Disable, StacLZS or
MPPC.
Multilink/QoS Default = Off
Enable the use of Multilink protocol (MPPC) on the link.
Incoming traffic Default = On
does not keep link
When enabled, the link is not kept up when the only traffic is incoming traffic.
up
LCP Echo Default = 6. Range = 0 to 99999 milliseconds.
Timeout (msecs)
When a PPP link is established, it is normal for each end to send echo packets to verify
that the link is still connected. This field defines the time between LCP echo packets. Four
missed responses in a row will cause the link to terminate.

Related links
L2TP Tunnel on page 559

IP Security Tunnel
IPSec allows data between two locations to be secured using various methods of sender
authentication and or data encryption. The use of IPSec requires entry of an IPSec Tunneling
license into the system at each end.
Related links
Tunnel on page 559
IPSec Main on page 562
Tunnel | IKE Policies (IPSec) on page 563
IPSec Policies on page 564

IPSec Main
Navigation: Tunnel | Main (IPSec)
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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 IP Security Tunnel

Field Descripition
Name Default = Blank.
A unique name for the tunnel. Once the tunnel is created, the name can be selected as a
destination for traffic in the IP Route table.
Local Configuration
The IP Address and IP Mask are used in conjunction with each other to configure and set the conditions for this
Security Association (SA) with regard to inbound and outbound IP packets.
IP Address The IP address or sub-net for the start of the tunnel.
IP Mask The IP mask for the above address.
Tunnel Endpoint The local IP address to be used to establish the SA to the remote peer. If left un-
IP Address configured, the system will use the IP address of the local interface on which the tunnel is
to be configured.

Remote Configuration
The IP Address and IP Mask are used in conjunction with each other to configure and set the conditions for this
Security Association (SA) with regard to inbound and outbound IP packets.
IP Address The IP address or sub-net for the end of the tunnel.
IP Mask The IP mask for the above address.
Tunnel Endpoint The IP address of the peer to which a SA must be established before the specified local
IP Address and remote addresses can be forwarded.

Related links
IP Security Tunnel on page 562

Tunnel | IKE Policies (IPSec)

Navigation: Tunnel | IKE Policies (IPSec)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Descripition
Shared Secret/ The password used for authentication. This must be matched at both ends of the tunnel.
Confirm
Password
Exchange Type Default = ID Prot
Aggressive provides faster security setup but does not hide the ID's of the communicating
devices. ID Prot is slower but hides the ID's of the communicating devices.
Encryption Default = 3DES CBC
Select the encryption method used by the tunnel. The option is:
• 3DES CBC
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Tunnel

Field Descripition
Authentication Default = SHA
The method of password authentication. The option is:
• SHA
DH Group Default = Group 1
Life Type Default = KBytes
Sets whether Life (below) is measured in seconds or kilobytes.
Life Range = 0 to 99999999.
Determines the period of time or the number of bytes after which the SA key is refreshed
or re-calculated.

Related links
IP Security Tunnel on page 562

IPSec Policies
Navigation: Tunnel | IKE Policies (IPSec)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Protocol Default = ESP
The options are:
• ESP (Encapsulated Security Payload)
• AH (Authentication Header, no encryption)
Encryption Default = DES3
Select the encryption method used by the tunnel. The option is:
• DES3
Authentication Default = HMAC SHA
The method of password authentication. The option is:
• HMAC SHA
Life Type Default = KBytes
Sets whether Life (below) is measured in seconds or kilobytes.
Life Determines the period of time or the number of bytes after which the SA key is refreshed
or re-calculated.

Related links
IP Security Tunnel on page 562

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Chapter 44: Auto Attendant (EVM)

• These settings are used for auto-attendants provided by embedded voicemail on an IP500
V2 control unit.
• For details of auto–attendants provided by Voicemail Pro on IP Office subscription systems,
see Auto Attendant (Voicemail Pro) on page 571.
For full details on configuration and operation of Embedded Voicemail auto-attendants, refer to the
IP Office Embedded Voicemail Installation.
Up to 40 auto-attendant services can be configured. Embedded voicemail services include auto-
attendant, callers accessing mailboxes to leave or collect messages and announcements to callers
waiting to be answered.
The IP500 V2 supports 2 simultaneous Embedded Voicemail calls by default but can be licensed for
up to 6. The licensed limit applies to total number of callers leaving messages, collecting messages
and or using an auto attendant.
In addition to basic mailbox functionality, embedded voicemail can also provide auto-attendant
operation. Each auto attendant can use existing time profiles to select the greeting given to callers
and then provide follow on actions relating to the key presses 0 to 9, * and #.
Time Profiles
Each auto attendant can use up to three existing time profiles, on each for Morning, Afternoon and
Evening. These are used to decide which greeting is played to callers. They do not change the
actions selectable by callers within the auto attendant. If the time profiles overlap or create gaps,
then the order of precedence used is morning, afternoon, evening.
Greetings
Four different greetings are used for each auto attendant. One for each time profile period. This is
then always followed by the greeting for the auto-attendant actions. By default a number of system
short codes are automatically created to allow the recording of these greetings from a system
extension. See below.
Actions
Separate actions can be defined for the DTMF keys 0 to 9, * and #. Actions include transfer to a
specified destination, transfer to another auto-attendant transfer to a user extension specified by the
caller (dial by number) and replaying the greetings.
• The Fax action can be used to reroute fax calls when fax tone is detected by the auto-
attendant.
• The Dial by Name action can be used to let callers specify the transfer destination.

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Auto Attendant (EVM)

Short Codes
Adding an auto attendant automatically adds a number of system short codes to assist in recording
the auto-attendant prompt. These use the Auto Attendant short code feature.
• System short codes (*81XX, *82XX, *83XX and *84XX) are automatically added for use
with all auto-attendants. These are used for morning, afternoon, evening and menu options
greetings respectively. These short codes use a Telephone Number of the form "AA:"N".Y"
where N is the replaced with the auto attendant number dialed and Y is 1, 2, 3 or 4 for the
morning, afternoon, evening or menu option greeting.
• To add a short code to call an auto-attendant, omit the XX part. For example, add the
short code *80XX/Auto Attendant/"AA:"N if internal dialed access to auto-attendants
is required.
• System short codes *800XX, *801XX, …, *809XX, *850XX, and *851XX are also
automatically added for recording prompts for any Page and Page actions. The codes
correspond to the key to which the action has been assigned; 0 to 9, * and # respectively.
These short codes use a Telephone Number of the form "AA:"N".00", …, "AA:"N".01",
"AA:"N".10" and "AA:"N".11" respectively.
Routing Calls to the Auto Attendant
The telephone number format AA:Name can be used to route callers to an auto attendant. It can be
used in the destination field of incoming call routes and telephone number field of short codes set to
the Auto Attend feature. Note however that when used with a short code it should be enclosed in
quotation marks, that is "AA:Name".
Related links
Auto Attendant on page 566
Actions on page 568

Auto Attendant
Navigation: Auto Attendant | Auto Attendant
These settings are used to define the name of the auto attendant service and the time profiles that
should control which auto attendant greetings are played.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Name Range = Up to 12 characters
This field sets the name for the auto-attendant service. External calls can be routed to the
auto attendant by entering AA:Name in the destination field of an Incoming Call Route.
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 Auto Attendant

Field Description
Maximum Default = 8 seconds; Range = 1 to 20 seconds.
Inactivity
This field sets how long after playing the prompts the Auto Attendant should wait for a
valid key press. If exceeded, the caller is either transferred to the Fallback Extension set
within the Incoming Call Route used for their call or else the caller is disconnected.
Enable Local Default = On.
Recording
When off, use of short codes to record auto-attendant prompts is blocked. The short
codes can still be used to playback the greetings.
Direct Dial-By- Default = Off.
Number
This setting affects the operation of any key presses in the auto attendant menu set to use
the Dial By Number action.
If selected, the key press for the action is included in any following digits dialed by the
caller for system extension matching. For example, if 2 is set in the actions to Dial by
Number, a caller can dial 201 for extension 201.
If not selected, the key press for the action is not included in any following digits dialed by
the caller for system extension matching. For example, if 2 is set in the actions to Dial by
Number, a caller must dial 2 and then 201 for extension 201.
Dial by Name Default = First Name/Last Name.
Match Order
Determines the name order used for the Embedded Voicemail Dial by Name function. The
options are:
• First then Last
• Last then First
AA Number This number is assigned by the system and cannot be changed. It is used in conjunction
with short codes to access the auto attendant service or to record auto attendant
greetings.
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Auto Attendant (EVM)

Field Description
Morning/ Each auto-attendant can consist of three distinct time periods, defined by associated time
Afternoon/ profiles. A greeting can be recorded for each period. The appropriate greeting is played to
Evening/Menu callers and followed by the Menu Options greeting which should list the available actions.
Options The options are:
• Time Profile The time profile that defines each period of auto-attendant operation.
When there are overlaps or gaps between time profiles, precedence is given in the order
morning, afternoon and then evening.
• Short code These fields indicate the system short codes automatically created to allow
recording of the time profile greetings and the menu options prompt.
• Recording Name: Default = Blank. Range = Up to 31 characters. This field appears
next to the short code used for manually recording auto-attendant prompts. It is
only used is using pre-recorded wav files as greeting rather than manually recording
greetings using the indicated short codes. If used, note that the field is case sensitive
and uses the name embedded within the wav file file header rather than the actual file
name.
This field can be used with all systems supporting Embedded Voicemail. The utility
for converting .wav files to the correct format is provided with Manager and can be
launched via File | Advanced | LVM Greeting Utility. Files then need to be manually
transferred to the Embedded Voicemail memory card. For full details refer to the IP
Office Embedded Voicemail Installation manual.

Related links
Auto Attendant (EVM) on page 565

Actions
Navigation: Auto Attendant | Actions
This tab defines the actions available to callers dependent on which DTMF key they press. To
change an action, select the appropriate row and click Edit. When the key is configured as
required click OK.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Key The standard telephone dial pad keys, 0 to 9 plus * and #.
The option Fax can be used for a transfer to the required fax destination and will then be
triggered by fax tone detection. If left as Not Defined, fax calls will follow the incoming call
routes fallback settings once the auto-attendant Maximum Inactivity Time set on the Auto
Attendant | Auto Attendant tab is reached.
Action
The following actions can be assigned to each key.
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 Actions

Field Description
Centrex Transfer Used to transfer the incoming call to an external telephone number defined in the Transfer
Number field.
Only supported for calls on Centrex analog trunks.
This option is only supported with Embedded Voicemail.
Dial by Name Callers are asked to dial the name of the user they require and then press #. The recorded
name prompts of matching users are then played back for the caller to make a selection.
The name order used is set by the Dial by Name Match Order setting on the Auto
Attendant tab. Note the name used is the user's Full Name if set, otherwise their User
Name is used. Users without a recorded name prompt or set to Exclude From Directory
are not included. For Embedded Voicemail in IP Office mode, users can record their name
by accessing their mailbox and dialing *05. For Embedded Voicemail in Intuity mode,
users are prompted to record their name when they access their mailbox.
Dial By Number This option allows callers with DTMF phones to dial the extension number of the user
they require. No destination is set for this option. The prompt for using this option should
be included in the auto attendant Menu Options greeting. A uniform length of extension
number is required for all users and hunt group numbers. The operation of this action is
affected by the auto attendant's Direct Dial-by-Number setting.
Normal Transfer Can be used with or without a Destination set. When the Destination is not set, this
action behaves as a Dial By Number action. With the Destination is set, this action
waits for a connection before transferring the call. Callers can hear Music on Hold.
Announcements are not heard.
Not Defined The corresponding key takes no action.
Park & Page The Park & Page feature is supported when the system Voicemail Type is designated
as Embedded Voicemail or Voicemail Pro. Park & Page is also supported on systems
where Modular Messaging over SIP is configured as the central voicemail system and
the local Embedded Voicemail provides auto attendant operation. The Park & Page
feature is an option in user mailboxes where a key is configured with the Park & Page
feature. When an incoming call is answered by the voicemail system and the caller dials
the DTMF digit for which Park & Page is configured, the caller hears the Park & Page
prompt. IP Office parks the call and sends a page to the designated extension or hunt
group. When Park & Page is selected in the Action drop-down box, the following fields
appear:
• Park Slot Prefix – the desired Park Slot prefix number. Maximum is 8 digits. A 0-9 will
be added to this prefix to form a complete Park Slot.
• Retry count – number of page retries; the range is 0 to 5.
• Retry timeout – provided in the format M:SS (minute:seconds). The range can be set in
15-second increments. The minimum setting is 15 seconds and the maximum setting is
5 minutes. The default setting is 15 seconds.
• Page prompt – short code to record the page prompt or upload the recorded prompt.
(Prompt can be uploaded to the SD card in the same way the AA prompts are).
Replay Menu Replay the auto-attendant greetings again.
Greeting
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Auto Attendant (EVM)

Field Description
Transfer Transfer the call to the selected destination. This is an unsupervised transfer, if the caller
is not answered they will be handled as per a direct call to that number.
Transfer to This action can be used to transfer calls to another existing auto attendant.
Attendant

Destination Sets the destination for the action.

Destination can be a user, a hunt group or a short code.
If the destination field is left blank, callers can dial the user extension number that they
require. Note however that no prompt is provided for this option so it should be included in
the auto attendant Menu Options greeting.
Consent Directive This field is used to control the addition of a consent value to the system’s SMDR output
and CTI call logging outputs. The intention is to allow the creation of auto-attendants
where having been prompts for consent to some issue, the caller’s response is included in
the system call logs. It can be set to the following:
• Not applicable: Set the consent value in the logging outputs to 0.
• Consent Denied: Set the consent value in the logging outputs to 6.
• Consent Given: Set the consent value in the logging outputs to 1.

Related links
Auto Attendant (EVM) on page 565

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Chapter 45: Auto Attendant (Voicemail Pro)

• These settings are used for the auto–attendants provided by Voicemail Pro on IP Office
subscription systems.
• For auto-attendants provided by embedded voicemail on an IP500 V2 control unit, see Auto
Attendant (EVM) on page 565.
Related links
Auto Attendant on page 571
Actions on page 575

Auto Attendant
Navigation: Auto Attendant | Auto Attendant
These settings are used to define the operation of the auto–attendant service whilst it waits for the
caller to select an option from the configured actions.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For IP Office R11.1 FP2 and higher, auto-attendants are also supported on systems that use
Voicemail Pro. However, the configuration of those auto-attendants is done using IP Office Web
Manager.
Auto-Attendant Settings
Field Description
Name Range = Up to 12 characters
The name for the auto-attendant. Set a name that acts as a reminder of the auto-
attendants role. The name is then also shown in other menus used to route calls to the
auto-attendant.
AA Number This number is automatically assigned by the system and cannot be changed. It is used in
conjunction with short codes to access the auto–attendant service or to record greetings.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
• IP500 V2 systems support up to 40 auto-attendants.
• IP Office Server Edition and Select systems support up to 100 auto-attendants.
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Auto Attendant (Voicemail Pro)

Field Description
Maximum Default = 8 seconds; Range = 1 to 20 seconds.
Inactivity
This value sets how long the attendant should wait for a response from the caller after
playing any current prompts.
• If the caller responds, their response is checked for a match to a configured action
without any further wait.
• Note that the caller can respond whilst the prompts are playing.
• If the timeout expires, the Menu Loop Count is checked to determine the next steps.
Name Match Default = Last then First
Order
This setting sets the name order used for the Dial By Name action if used.
Direct By Number Default = No
This setting affects the operation keys set to the Dial By Number action.
• If enabled: The caller’s key press to select the action is included in the digits they dial
for a extension match. For example, if menu key 2 is used for the action, a caller can
dial 2 and then 01 for extension 201.
• If not enabled: The caller’s key press to select the action is not included in the digits
they dial for extension match. For example, if menu key 2 is used for the action, a caller
must dial 2 and then 201 for extension 201.
Direct By Default = No
Conference
This setting affects the operation keys set to the Dial By Conference action.
• If enabled: The caller’s key press to select the action is included in the digits they dial
for a conference match. For example, if menu key 3 is used for the action, a caller can
dial 3 and then 01 for conference 301.
• If not enabled: The caller’s key press to select the action is not included in the digits
they dial for a conference match. For example, if menu key 3is used for the action, a
caller must dial 3 and then 301 for conference 301.
Enable Local Default = Yes
Recording
When off, use of short codes to record auto-attendant prompts is blocked. The short
codes can still be used to playback the greetings.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
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 Auto Attendant

Field Description
Speech AI Default = Off
This option is only available on subscription mode systems. It sets whether the auto-
attendant supports text-to-speech and automatic speech recognition features.
• When off, the auto-attendant does not support any text-to-speech and speech
recognition features.
- The language used for any prompts provided by the system is determined from the
call settings. See Google TTS Prompt Language on page 881.
• When set to a specific language, the auto-attendant supports text-to-speech and speech
recognition features in that language.
- It also uses that language for all system prompts it provides regardless of the locale
call settings the system has associated with the call.
Speech Voice This setting is available when Speech AI is set to a specific language. It allows selection
of a particular voice used for any text-to-speech features.
See Text-to-Speech (TTS) Prompts on page 881.

Greeting and Announcement Settings

When a caller reaches an auto-attendant, they first hear the attendant's current greeting (if any)
and then the attendant's menu announcement.
• The greeting used is the first one (from up to 3 defined greetings) for which the greeting's
associated time profile is currently active. This allows you to define greetings for different
times of day (for example “Good Morning”, “Good Afternoon” and “Sorry, we are currently
closed”) or different greetings for business and non-business days.
• The menu announcement should contain the instructions for the caller regarding the keys
they can press and other actions.
• Each time a caller goes round the auto-attendant loop, they can respond (with key presses or
speech) whilst any greeting and announcement menu prompt is being played.
Field Description
Optional Greeting Up to 3 greetings can be defined using the Add Greeting button.
1
• Each greeting requires an associated time profile.
Optional Greeting
2 - Time Profile: Default = Off (Greeting not used).

Optional Greeting • If Off, the greeting is not used.

3 • The greeting is only used when defined by its associated time profile.
• When multiple greetings are defined, the first one that has an active time profile, in the
order 1 to 3, is used as the current greeting.
• If no greetings is currently active according to its time profile, then no greeting is played.
• If a greeting is no longer required, it can be deleted by clicking on the adjacent icon.
• After playing any greeting, the system always then plays the menu announcement.
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Auto Attendant (Voicemail Pro)

Field Description
Menu The menu announcement should contain the instructions for callers about the actions they
Announcement can perform. For example; “Press 1 for reception. Press 2 for sales, ...””
It is used as follows:
• When a call first reaches the auto-attendant, it is played to the caller after whichever
greeting is currently active.
• If the Menu Loop Count is not zero, it is played again at the start of each repeat loop.
• The caller can respond by pressing a key whilst the announcement is being played. On
subscription mode systems, if Speech AI is enabled they can also respond by speaking
whilst the announcement is played.
• After the announcement is played, the auto-attendant waits for a response for the time
set by the Maximum Inactivity setting.
Menu Loop Count Default = 0 (No Repeat)
This setting sets the number of times the auto-attendant will repeat the Menu
Announcement and then wait for a valid response.
If the caller does not respond or their response is not matched to an action:
• If 0, the default, they hear the No Match Prompt prompt and the Fallback Action
setting is used.
• If non-zero but the number of repeat loops has not been reached, they hear the No
Match Prompt and then the Menu Announcement again and the auto-attendant waits
for a response again.
• If non-zero and the number of repeat loops has been reached, they hear the No Match
Prompt prompt and the Fallback Action setting is used.
No Match Prompt This prompt is heard when the caller does not respond in time or if their response does
not match a configured action. For example; “Sorry, no response was recognized.”
• Note that this prompt is also heard by callers who are about to be redirected to the
Fallback Action. Therefore a prompt like “"Please try again"” would not be appropriate.

The following settings are common to the menu announcement, greetings and error message.
The greetings and announcements can be recorded from the phone, use an uploaded file or be
provided by text-to-speech. Whichever method was last used or configured overrides any previous
prompt.
Field Description
Dial To Record Default = Automatically assigned. Not changeable.
Greeting
This field indicates the short code that can be dialed in order to record the greeting from
an internal extension.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
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 Actions

Field Description
Audio Output Default = Audio File
The field sets the current method used to provide the prompt used for the greeting or
announcement. Clicking on the current value allows you to see its current settings and to
change them or to change the recording method.
• Audio File (wav) – Provide the prompt using a pre-recorded audio file.
See Using Pre-Recorded Prompt Files on page 907.

Note:
Use IP Office Web Manager to upload the .wav file.
• Text To Speech – Provide the prompt using the text-to-speech service. This option
is only available on subscription mode systems with Speech AI enabled and set to a
specific language.
See Recording Auto-Attendant Prompts Using Text-to-Speech on page 908.

Related links
Auto Attendant (Voicemail Pro) on page 571

Actions
Navigation: Auto Attendant | Actions
This tab defines the actions available to callers dependent on which DTMF key they press or, on
subscription mode systems, based on automatic speech recognition of keywords. To change an
action, click on the appropriate button.
The Fallback Action action applied is the user does not make a recognized choice is configured
separately through the No Match Prompt prompt settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Settings: Keys/Events
The following actions can be assigned to the selected keys.
Action Description
0 to 9, *, # These keys correspond to the standard telephone dial pad key. Clicking on a key allows
configuration of its settings.
Fax If configured, the Fax option is used when the system detects fax tone.
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Auto Attendant (Voicemail Pro)

Action Description
Fallback Action Default = Drop Call
This option is used when the number of times the auto-attendant has waited for a valid
response from the caller has exceeded the Menu Loop Count. It is preceded by the No
Match Prompt and then the configured action is performed.
All actions are supported except Park & Page, Replay Menu Greeting, Speak By Name
and Speak By Number
You can choose whether to mention this option in the Menu Announcement. For
example, if set to transfer to your receptionist, add “... or wait to for our operator.”
Menu The menu announcement should contain the instructions for callers about the actions they
Announcement can perform. For example; “Press 1 for reception. Press 2 for sales, ...””
It is used as follows:
• When a call first reaches the auto-attendant, it is played to the caller after whichever
greeting is currently active.
• If the Menu Loop Count is not zero, it is played again at the start of each repeat loop.
• The caller can respond by pressing a key whilst the announcement is being played. On
subscription mode systems, if Speech AI is enabled they can also respond by speaking
whilst the announcement is played.
• After the announcement is played, the auto-attendant waits for a response for the time
set by the Maximum Inactivity setting.

Settings: Key Actions

Action Description
Not configured Perform no action.
Dial By Allow the caller to dial the conference ID they require.
Conference
See Dial By Conference on page 892.
Dial By Name Prompt the caller to dial the name of the user they require.
See Dial By Name on page 893.
Dial By Number Allow the caller to dial the extension number they require.
See Dial By Number on page 895.
Leave Message Redirect the caller a specified mailbox to leave a message.
See Leave Message on page 896.
Supervised Transfer the caller to the specified extension number.
Transfer
See Supervised Transfer on page 897.
Park & Page Park the call and make an announcement to the a specified group.
See Park & Page on page 898.
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 Actions

Action Description
Replay Menu Replay the auto-attendant’s menu announcement.
Greeting
See Replay Menu on page 900.
Unsupervised Transfers the caller to the specified extension number.
Transfer
See Unsupervised Transfer on page 903.
Transfer To Auto Transfers the caller to another auto–attendant.
Attendant
See Transfer to Auto Attendant on page 904.
Speak By Name Allow the caller to select from listed names using speech.
See Speak By Name on page 901.
Speak By Number Allow the caller to speak the extension number required.
See Speak By Number on page 902.
Destination The destination depends on the action:
• Leave Message, Supervised Transfer and Unsupervised Transfer – Use the drop-
down to select the target extension.
• Transfer To Auto Attendant – Use the drop-down to select another existing auto-
attendant.
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Auto Attendant (Voicemail Pro) on page 571

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Chapter 46: Authorization Codes

Navigation: Authorization Codes

When a user dials an external number that matches a short code set to Force Authorization Code,
the IP Office system will prompt the user to enter their associated Authorization Code before
allowing the call to continue.
Valid/invalid authorization code entry is recorded in the SMDR output. The code used is not
recorded.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Note:
For Release 9.1 and higher, you can no longer associate Authorization Code entries with User
Rights. Authorization Code configured in that way are removed during the upgrade.
Field Description
Authorization Code Range = Up to 12 digits.
The digits used for the authorization code. Each code must be unique. Wildcards are
not usable with authorization codes.
User This field is used to select a user with which the authorization code is associated. The
authorization code can then be used to authorize calls made by that user.

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Chapter 47: User Rights

Additional configuration information

This section provides the User Rights field descriptions. For additional configuration information, see
Configuring User Rights on page 740.
• For systems using record consolidation, you can only add and edit this type of record at the
solution level. The record is then automatically copied to each IP Office system in the network.
User Rights act as templates for selected user settings. The settings of a user rights template are
applied to all users associated with that template. The use of a template can also be controlled by a
time profile to set when the template is used for a particular user.
Related links
User on page 579
Short Codes on page 580
Button Programming on page 581
Telephony on page 581
User Rights Membership on page 586
Voicemail on page 587
Forwarding on page 588

User
Navigation: User Rights | User
Used to set and lock various user settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Name The name for the user rights . This must be set in order to allow the user rights to be
selected within the User Rights drop down list on the User | User tab of individual users.
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User Rights

Field Description
Application Default = Off.
Servers Group
Set to On if the IP Office system is deployed in an IP Office Contact Center solution or an
Avaya Contact Center Select solution.
Only one user rights record can be configured to be the Application Servers Group. If it is
set on any one group then the control is disabled on all other groups.
Locale Default = Blank
Sets and locks the language used for voicemail prompts to the user, assuming the
language is available on the voicemail server. On a digital extension it also controls the
display language used for messages from the system to the phone. See Avaya IP Office
Locale Settings.
Priority Default = 5, Range 1 (Lowest) to 5 (Highest)
Sets and locks the user's priority setting for least cost routing.
Do Not Disturb Default = Off Sets and locks the user's DND status setting.

Related links
User Rights on page 579

Short Codes
Navigation: User Rights | Short Codes
Used to set and lock the user's short code set. The tab operates in the same way as the User |
Short Codes tab. User and User Rights short codes are only applied to numbers dialed by that
user. For example they are not applied to calls forwarded via the user.
Warning:
User dialing of emergency numbers must not be blocked. If short codes are edited, the users
ability to dial emergency numbers must be tested and maintained.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Related links
User Rights on page 579

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 Button Programming

Button Programming
Navigation: User Rights | Button Programming
This tab is used to set and lock the user's programmable button set. When locked, the user cannot
use Admin or Admin1 buttons on their phone to override any button set by their user rights.
Buttons not set through the user rights can be set through the user's own settings. When Apply
user rights value is selected, the tab operates in the same manner as the User | Button
Programming tab.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Adding Blank Buttons
There are scenarios where users are able to program their own buttons but you may want to
force certain buttons to be blank. This can be done through the user's associated User Rights as
follows:
1. Assign the action Emulation | Inspect to the button. This action has no specific function.
Enter some spaces as the button label.
2. When pressed by the user, this button will not perform any action. However it cannot be
overridden by the user.
Related links
User Rights on page 579

Telephony
Navigation: User Rights | Telephony
Allows various user telephony settings to be set and locked. These match settings found on the
User | Telephony tab.
Related links
User Rights on page 579
Call Settings on page 581
Supervisor Settings on page 582
Multi-line Options on page 584
Call Log on page 585

Call Settings
Navigation: User Rights | Telephony | Call Settings

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User Rights

Additional configuration information

For additional information on ring tones, see Ring Tones on page 655.
Configuration settings
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
No Answer Time Default = Blank (Use system setting). Range = 6 to 99999 seconds.
Sets how long a call rings the user before following forwarded on no answer if set or going
to voicemail. Leave blank to use the system default setting.
Transfer return Default = Blank (Off), Range 1 to 99999 seconds.
Time (secs)
Sets the delay after which any call transferred by the user, which remains unanswered,
should return to the user if possible.
Wrap up Time Default = 2 seconds, Range 0 to 99999 seconds.
(secs)
Specifies the amount of time after ending one call before another call can ring. You may
wish to increase this in a "call center" environment where users may need time to log call
details before taking the next call. It is recommended that this option is not set to less than
the default of 2 seconds. 0 is used for immediate ringing.
Call waiting Default = Off
on/Enable call
For users on phones without appearance buttons, if the user is on a call and a second
waiting
call arrives for them, an audio tone can be given in the speech path to indicate a waiting
call (the call waiting tone varies according to locale). The waiting caller hears ringing rather
than receiving busy. There can only be one waiting call, any further calls receive normal
busy treatment. If the call waiting is not answered within the no answer time, it follows
forward on no answer or goes to voicemail as appropriate. User call waiting is not used for
users on phones with multiple call appearance buttons.
Busy on held/ Default = Off
Enable busy on
If on, when the user has a call on hold, new calls receive busy tone (ringing for incoming
Held
analog call) or are diverted to voicemail if enabled, rather than ringing the user. Note this
overrides call waiting when the user has a call on hold. Not supported (should be set to off)
for users with call appearance buttons.

Related links
Telephony on page 581

Supervisor Settings
Navigation: User Rights | Telephony | Supervisor Settings
Additional configuration information
Off-Switch Transfer Restriction

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 Telephony

Call Barring
Configuration settings
These settings relate to user features normally only adjusted by the user's supervisor.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Can Intrude Default = Off
If enabled, the user can perform is allowed to perform a range of action on other user's
calls. For example: Call Intrude, Call Listen, Call Steal and Dial Inclusion. See Call
Intrusion on page 716.
Cannot be Default = On
Intruded
If checked, this user's calls cannot be interrupted or acquired by users who have Can
Intrude enabled. This setting also affects whether other users can use their appearance
buttons to bridge into a call to which this user has been the longest present user.
Deny Auto Default = Off.
Intercom Calls
When enabled, any automatic intercom calls to the user's extension are automatically
turned into normal calls.
Force Login Default = Off
If checked, the user must log in using their Login Code to use an extension. For example,
if Force Login is ticked for User A and user B has logged into A's phone, after B logs off A
must log back. If Force Login was not ticked, A would be automatically logged back in.
Force Account Default = Off
Code
If checked, the user must enter a valid account code to make an external call.
Inhibit Off-Switch : Default = Off
Forward/Transfer
When enabled, this setting stops the user from transferring or forwarding calls externally.
Note that all user can be barred from forwarding or transferring calls externally by the
System | Telephony | Telephony | Inhibit Off-Switch Forward/Transfers setting.
Outgoing Call Default = Off
Bar
When set, bars the user from making external calls.
Coverage Group Default = <None>
If a group is selected, the system will not use voicemail to answer the users unanswered
calls. Instead the call will continue ringing until either answered or the caller disconnects.
For external calls, after the users no answer time, the call is also presented to the users
who are members of the selected Coverage Group. For further details refer to Coverage
Groups.
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User Rights

Field Description
ICR Agent Applicable for Integrated Contact Reporter
Default = Off
Enable to configure user right members as ICR agents. Any user configured to use the
user right becomes an ICR agent.
If enabled, it also activates the After Call Work related fields.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Automatic After Applicable for Integrated Contact Reporter
Call Work
Default = Off
If enabled, all ICR agents of the user right go into After Call Work (ACW) at the end of an
ICR and non-ICR hunt group call to indicate that they are busy with post-call processing
activity. During the ACW state, they are not sent any hunt group calls.
For more information about configuring ACW, see Administering Avaya IP Office™
Platform Integrated Contact Reporter.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Can Control after Applicable for Integrated Contact Reporter
Call Work
Default = Off
If enabled, the ICR agents in the user right can extend the currently active After Call Work
time indefinitely.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
After Call Work Applicable for Integrated Contact Reporter
Time
Default = The value in this field is populated from the Default After Call Work Time field
located at System | Contact Center.
The time after a call when an agent is busy and unable to deal with hunt group calls.
Change the value if you want to specify ACW time for this all ICR agents in the user right
to be different from the system default.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Related links
Telephony on page 581

Multi-line Options
Navigation: User Rights | Telephony | Multi-line Options

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 Telephony

Additional configuration information

For additional configuration information, see Appearance Button Operation on page 1159.
Configuration settings
Multi-line options are applied to a user's phone when the user is using an Avaya phones which
supports appearance buttons (call appearance, line appearance, bridged and call coverage).
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Individual Default = 10 seconds, Range 1 to 99999 seconds.
Coverage Time
This function sets how long the phone will ring at your extension before also alerting at
(secs)
any call coverage users. This time setting should not be equal to or greater than the No
Answer Time.

Related links
Telephony on page 581

Call Log
Navigation: User Rights | Telephony | Call Log
The IP Office stores a centralized call log for each user:
• The user's centralized call log is stored by the IP Office system on which the user is
configured. If the user is logged in on another system, new call log records are sent to
the user's home IP Office system, but using the time and date on the IP Office system where
the user is logged in.
• Each user's centralized call log can contain 30 (IP500 V2) or 60 (Server Edition) call records.
Each new call record replaces the oldest previous record when it reaches the limit.
• By default, the centralized call log is displayed on Avaya phones with a fixed Call Log
or History button (1400, 1600, 9500, 9600, J100 Series) and in the one-X Portal, Avaya
Workplace Client, and IP Office User Portal applications.
• The centralized call log moves with the user as they log onto different phones and IP Office
applications.
• The missed call count is updated per unique caller, not per call.

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User Rights

Field Description
Centralized Call Default = System Default (On)
Log
This setting allows the use of centralized call logging to be enabled or disabled on a per
user basis. The default is to match the system setting System | Telephony | Call Log |
Default Centralized Call Log On.
The other options are On or Off for the individual user. If off is selected, the call log shown
on the users phone is the local call log stored by the phone.
Delete Default = 00:00 (Never).
records after
(hours:minutes) If a time period is set, records in the user's call log are automatically deleted after this
period.
Groups Default = System Default (On).
This section contains a list of hunt groups on the system. If the system setting System
| Telephony | Call Log | Log Missed Huntgroup Calls has been enabled, then missed
calls for those groups selected are shown as part of the users call log. The missed calls
are any missed calls for the hunt group, not just group calls presented to the user and not
answered by them.

Related links
Telephony on page 581

User Rights Membership

Navigation: User Rights | User Rights Membership
The tabs display the users associated with the user rights and allows these to be changed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Members of this This tab indicates those users associated with the user rights. If the user has an
User Rights associated Working hours time profile, their association to the user rights applies only
during the periods defined by the time profile. If the user does not have an associated
Working hours time profile, they are associated with the user rights at all times.
Members when This tab indicates those users associated with the user rights outside the time periods
out of service defined by their Working hours time profile. The Members when out of service tab is not
populated unless there are time profiles available within the configuration.

Related links
User Rights on page 579

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 Voicemail

Voicemail
Navigation: User Rights | Voicemail
Display the users associated with the user rights and allows these to be changed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Voicemail On Default = On
When on, the mailbox is used by the system to answer the user's unanswered calls or
calls when the user's extension returns busy. Note that selecting off does not disable use
of the user's mailbox. Messages can still be forward to their mailbox and recordings can
be placed in it. The mailbox can also still be accessed to collect messages.
Voicemail Default = Off
Ringback
When enabled and a new message has been received, the voicemail server calls the
user's extension to attempt to deliver the message each time the telephone is put down.
Voicemail will not ring the extension more than once every 30 seconds.
DTMF Breakout
When a caller is directed to voicemail to leave a message, they can be given the option to be transferred to
a different extension. The greeting message needs to be recorded telling the caller the options available. The
extension numbers that they can be transferred to are entered in the fields below. These system default values
can be set for these numbers and are used unless a different number is set within these user settings.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for Enterprise Branch with CS 1000 deployments) is configured as the central
voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant operation. The
Park & Page feature allows a call to be parked while a page is made to a hunt group or extension. This feature
can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
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User Rights

Field Description
Reception/ The number to which a caller is transferred if they press 0while listening to the mailbox
Breakout (DTMF greeting rather than leaving a message (*0 on Embedded Voicemail in IP Office mode).
0)
For voicemail systems set to Intuity emulation mode, the mailbox owner can also access
this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave Mail
action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success results
connections depending on whether the caller pressed 0 before or after the record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0) setting.
When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
• Paging Number – displays a list of hunt groups and users (extensions). Select a hunt
group or extension to configure this option.
• Retries – the range is 0 to 5. The default setting is 0.
• Retry Timeout – provided in the format M:SS (minute:seconds). The range can be set
in 15-second increments. The minimum setting is 15 seconds and the maximum setting
is 5 minutes. The default setting is 15 seconds
Breakout (DTMF The number to which a caller is transferred if they press 2while listening to the mailbox
2) greeting rather than leaving a message (*2 on Embedded Voicemail in IP Office mode)
Breakout (DTMF The number to which a caller is transferred if they press 3while listening to the mailbox
3) greeting rather than leaving a message (*3 on Embedded Voicemail in IP Office mode).

Related links
User Rights on page 579

Forwarding
Navigation: User Rights | Forwarding
Additional configuration information
For additional configuration information, see DND, Follow Me, and Forwarding on page 744.
Configuration settings
Display the users associated with the user rights and allows these to be changed.
These settings are mergeable.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.

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 Forwarding

Field Description
Block Forwarding
Enable Block Default = Off.
Forwarding
When enabled, call forwarding is blocked.
The following actions are blocked:
• Follow me
• Forward unconditional
• Forward on busy
• Forward on no answer
• Call Coverage
• Hot Desking
The following actions are not blocked:
• Do not disturb
• Voicemail
• Twinning

Related links
User Rights on page 579

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Chapter 48: Conference

These settings are used to define the operation of a system meet-me conferences. These
are supported on subscription mode systems. For more details, see System Conferences on
page 927.
Field Description
Conference ID Range = Up to 15 digits.
This ID is shown in the destination list for auto-attendant actions and incoming call routes.
The ID can also be used with short code and programmable button features in order to
access the conference.
• Do not enter a number that matches a user’s extension number. Doing so will override
that user’s personal meet-me conference facility.
• It is advisable not to use conference ID’s that are near the range that may be in use for
ad-hoc conferences as above (100 plus). Once a conference ID is in use by an ad-hoc
conference, it is no longer possible to join the conference using the various conference
meet me features.
Name This is a short name to help indicate the system conferences intended use. For example,
“Sales Team”.
Moderator List Optional. Default = No moderators.
List the internal users who are moderators for this system conference, up to a maximum of
8 moderators. When set:
• The conference Hold Music is played to other participants when there is no moderator in
the conference.
• These user’s do not need to enter a PIN in order to access the conference.
• Listed users using the User Portal application can view the conference PIN details.
In addition:
• Other participants, including external participants, can become moderators by entering
the Moderator Pin when they join the conference.
• Conferences with no defined moderators (blank Moderator List and no Moderator Pin)
start immediately any caller joins and can have recording started/stopped by any internal
user.
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Field Description
Delegate Pin Optional. Range = Up to 30 digits.
If set, the system will prompt callers, other than those in the Moderator List list, to enter a
PIN before it allows them to join the conference.
The system allows 3 PIN entry attempts before disconnecting the caller.
Moderator Pin Optional. Range = Up to 30 digits.
If set, callers who enter this PIN rather than the Delegate Pin are added to the conference
as a moderator. This allows moderators who are not in the Moderator List including
external callers. Note however that external callers will not be able to access moderator
controls other than starting/stopping the conference by their presence.
Hold Music Default = Tone
If the conference has been configured with moderators, this music is played to other
participants who join the conference when no moderator is present. The music is also
played if any present moderators leave the conference.
• Tone – Play repeated system tones to participants whilst waiting for a conference
moderator.
• System – Use the system’s default music-on-hold. This option is only shown in a music-
on-hold file has been uploaded.
• If other music sources have been configured, they can also be selected from the drop-
down list.
Before the hold music is played, participants will hear a prompt informing them of the
reason for hearing the music.
Speech AI Default = Same as system
On subscription systems, this and other text-to-speech options are available if the System
| Voicemail setting for Google Speech AI is enabled.
• If set to Same as System, the settings of the System | Voicemail form are used for TTS
prompts.
• If set to Custom, the Language and Voice fields below can be used.
Language Default = Matches the system locale.
Set the language used for prompts provided by the system for the system conference.
Voice Sets the voice to be used with the speech language. The number of voices available varies
depending on the speech language selected.
Recording Type Default = Manual
Sets the method by which recording of the system conference is controlled:
• Manual – Recording can be started/stopped by moderators.
• Private – No recording allowed.
• Automatic – Automatically start recording the conference when started. The recording
can be stopped/restarted by moderators.
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Conference

Field Description
Recording Default = Conference Mailbox
Destination
Sets the destination for system conference recordings. Note that the selected option may
also affect the maximum recording length:
• Conference Mailbox - Place calls into a standard group mailbox, using the conference
ID as the mailbox number. Maximum recording length 60 minutes. Message waiting
indication and visual voice access can be configured by adding C<conference ID> to a
user’s source numbers.
• Conference VRL - Transfer the conference recordings to the systems VRL application
(on subscription systems, set by the System > System > Media Archival Solution
setting). Maximum recording length 5 hours.
Meeting Arrival Default = Off
Announcement
If enabled, the system plays this prompt to callers before they join the conference. If
conference PIN codes have been defined, it is played before the request to the caller to
enter their PIN code.
• Audio Output – Use an uploaded audio file. See .The file must be a .wav file in Mono
PCM 16-bit format, either 8, 16 or 22KHz. Maximum length 10 minutes. To upload a file
click on Upload and select the required file. Alternatively, click and drag the file onto the
download box.

Note:
Use IP Office Web Manager to upload the .wav file.
• Text-to-Speech – Use a prompt generated using TTS. Up to 200 characters.

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Chapter 49: ARS

ARS (Alternate Route Selection) replaces LCR (Least Cost Routing) used by previous releases of IP
Office. It also replaces the need to keep outgoing call routing short codes in the system short codes.

ARS
Navigation: ARS | ARS
Additional configuration information
This section contains the configuration settings for Alternate Route Selection. For additional
configuration information, see Configuring ARS on page 691
Configuration settings
Each ARS form contains short codes which are used to match the result of the short code that
triggered use of the ARS form, ie. the Telephone Number resulting from the short code is used
rather than the original number dialed by the user.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
ARS Route ID The default value is automatically assigned. Range = 0 to 99999.
For most deployments, do not edit this field.
For those conditions where it is necessary to edit this field, the value must be unique
within ARS and within the line Outbound Group IDs.
Route Name Default = Blank. Range = Up to 15 characters.
The name is used for reference and is displayed in other areas when selecting which
ARS to use.
Dial Delay Time Default = System. Range = 1 to 30 seconds.
This settings defines how long ARS should wait for further dialing digits before assuming
that dialing is complete and looking for a short code match against the ARS form short
codes. When set to System, the system setting System | Telephony | Telephony | Dial
Delay Time is used.
Table continues…

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ARS

Field Description
Secondary Dial Defaults = Off.
Tone
When on, this setting instructs the system to play secondary dial tone to the user. The
tone used is set by the field below.
The tone used is set as either System Tone (normal dial tone) or Network Tone
(secondary dial tone). Both tone types are generated by the system in accordance
with the system specific locale setting. Note that in some locales normal dial tone and
secondary dial tone are the same.
When Secondary Dial Tone is selected, the ARS form will return tone until it receives
digits with which it can begin short code matching. Those digits can be the result of user
dialing or digits passed by the short code which invoked the ARS form. For example with
the following system short codes:
In this example, the 9 is stripped from the dialed number and is not part of the telephone
number passed to the ARS form. So in this case secondary dial tone is given until the
user dials another digit or dialing times out.
• Code: 9N
• Telephone Number: N
• Line Group ID: 50 Main
In this example, the dialed 9 is included in the telephone number passed to the ARS form.
This will inhibit the use of secondary dial tone even if secondary dial tone is selected on
the ARS form.
• Code: 9N
• Telephone Number: 9N
• Line Group ID: 50 Main
Check User Call Default = Off
Barring
If enabled, the dialing user's Outgoing Call Bar setting and any user short codes set to
the function Barred are checked to see whether they are appropriate and should be used
to bar the call.
Description Default = Blank. Maximum 31 characters.
You can use this field to enter a description for the configuration entry. The description is
not used elsewhere.
In Service: Default = On
This field is used to indicate whether the ARS form is in or out of service. When out of
service, calls are rerouted to the ARS form selected in the Out of Service Route field.
Short codes can be used to take an ARS form in and out of service. This is done using
the short code features Disable ARS Form and Enable ARS Form and entering the ARS
Route ID as the short code Telephone Number value.
Out of Service Default = None.
Route
This is the alternate ARS form used to route calls when this ARS form is not in service.
Table continues…

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 ARS

Field Description
Time Profile Default = None.
Use of a ARS form can be controlled by an associate time profile. Outside the hours
defined within the time profile, calls are rerouted to an alternate ARS form specified in the
Out of Hours Route drop-down. Note that the Time Profile field cannot be set until an Out
of Hours Route is selected.
Out of Hours Default = None.
Route
This is the alternate ARS form used to route calls outside the hours defined within the
Time Profile selected above.
Short Codes Short codes within the ARS form are matched against the "Telephone Number" output by
the short code that routed the call to ARS. The system then looks for another match using
the short codes with the ARS form.
Only short codes using the following features are supported within ARS: Dial, Dial
Emergency, Dial Speech, Dial 56K, Dial64K, Dial3K1, DialVideo, DialV110, DialV120
and Busy.
Multiple short codes with the same Code field can be entered so long as they have
differing Telephone Number and or Line Group ID settings. In this case when a match
occurs the system will use the first match that points to a route which is available.
Alternate Route Default = 3. Range = 1 (low) to 5 (high).
Priority
If the routes specified by this form are not available and an Alternate Route has been
specified, that route will be used if the users priority is equal to or higher than the value
set here. User priority is set through the User | User form and by default is 5. If the users
priority is lower than this value, the Alternate Route Wait Time is applied. This field is
grayed out and not used if an ARS form has not been selected in the Alternate Route
field.
If the caller's dialing matches a short code set to the Barred function, the call remains at
that short code and is not escalated in any way.
Alternate Route Default = 30 seconds. Range = Off, 5 to 60 seconds.
Wait Time
If the routes specified by this form are not available and an Alternate Route has been
specified, users with insufficient priority to use the alternate route immediately must wait
for the period defined by this value. During the wait the user hears camp on tone. If during
that period a route becomes available it is used. This field is grayed out and not used if an
ARS form has not been selected in the Alternate Route field.
Alternate Route Default = None.
This field is used when the route or routes specified by the short codes are not available.
The routes it specifies are checked in addition to those in this ARS form and the first route
to become available is used.

Cause Codes and ARS

ARS routing to digital trunks can be affected by signalling from the trunk.
The following cause codes cause ARS to no longer target the line group (unless it is specified by
an alternate ARS route). The response to cause codes received from the line is as follows.

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ARS

Code Cause Code

1 Unallocated Number.
2 No route to specific transit network/(5ESS) Calling party off hold.
3 No route to destination./(5ESS) Calling party dropped while on hold.
4 Send special information tone/(NI-2) Vacant Code.
5 Misdialed trunk prefix.
8 Preemption/(NI-2) Prefix 0 dialed in error.
9 Preemption, cct reserved/ (NI-2) Prefix 1 dialed in error.
10 (NI-2) Prefix 1 not dialed.
11 (NI-2) Excessive digits received call proceeding.
22 Number Changed.
28 Invalid Format Number.
29 Facility Rejected.
50 Requested Facility Not Subscribed.
52 Outgoing calls barred.
57 Bearer Capability Not Authorized.
63 Service or Option Unavailable.
65 Bearer Capability Not Implemented.
66 Channel Type Not Implemented.
69 Requested Facility Not Implemented.
70 Only Restricted Digital Information Bearer Capability Is Available.
79 Service Or Option Not Implemented.
88 Incompatible.
91 Invalid Transit Network Selection.
95 Invalid Message.
96 Missing Mandatory IE.
97 Message Type Nonexistent Or Not Implemented.
98 Message Not Implemented.
99 Parameter Not Implemented.
100 Invalid IE Contents.
101 Msg Not Compatible.
111 Protocol Error.
127 Interworking Unspecified.

Stop ARS The following cause codes stop ARS targeting completely.

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 ARS

Code Cause Code

17 Busy.
21 Call Rejected.
27 Destination Out of Order.

No Affect All other cause codes do not affect ARS operation.

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Chapter 50: Location

Location records can be used to identify where particular extensions are physically located and to
apply settings that need to differ from that location.
• When Locations have been defined, you must configure the system with one of those
locations.
• For systems using record consolidation, you can only add and edit this type of record at the
solution level. The record is then automatically copied to each IP Office system in the network.
For additional configuration information, see:
• Emergency Call on page 652
• Configuring Call Access Control on page 709
• Preventing Toll Bypass on page 707
Defaults
By default, new lines and extensions are assigned the same location as set for their host IP Office
system. However, their location setting can be changed individually. For IP extensions, the location
can also be set to automatically by matching the IP extension's current IP address to the address
settings of an existing location.
Networked Configurations
In networked IP Office configurations, each location entry and its settings are automatically
replicated in the configuration of all IP Office systems in the network. The exception is the
Emergency ARS setting which can be configured separately for the same location entry on each
system.
Related links
Location on page 598
Address on page 601

Location
Navigation: Location | Location
Locations allows you to apply a range of common settings to systems, extensions and IP lines
that are in the same location. For example, each location can define the timezone settings to be
applied to extensions in that location. See Using Locations on page 617.

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 Location

Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Location Name Default = Blank.
A meaningful location name, clearly identifying the location. The location name is included
in system alarms for emergency calls. It is also shown on J189 phones with an emergency
view button.
Location ID Default = Based on existing configured locations, the next incremental value is assigned.
This field is read only. For DECT R4, this value can be entered into the configuration of
a base station in order to associate emergency calls made by extensions using that base
station with location emergency ARS and address settings. Refer to the IP Office DECT
R4 Installation manual.
Subnet Address Default = Blank.
The IP address associated with this location. The subnet where this IP address resides
must be unique across all configured locations. Overlapping IP address ranges between
locations will cause extensions to use the first match found which may not be the correct
location.
Subnet Mask Default = Blank.
The subnet mask for this IP address.
Emergency ARS Default = None.
This setting sets which ARS (Alternate Route Selection) entry on the system should
be used to route emergency calls from location. Refer to the IP Office Emergency Call
Configuration manual.
When the dialing on an extension associated with the location matches a Dial Emergency
shortcode, this setting overrides the Line Group ID setting of the shortcode.
Fallback System Default = No override.
The drop down list contains all configured IP Office Lines and the associated IP Office
system. The group of extensions associated with this location can fallback to the alternate
system selected.

Call Admission Control

The call admission control (CAC) settings allow the number of calls on IP trunks between locations
to be controlled. See Configuring Call Admission Control on page 709.
Field Description
Total Maximum Default = Unlimited. Range = 1 - 99, Unlimited.
Calls
Limit of all calls to or from other configured locations and the cloud.
External Default = Unlimited. Range = 1 - 99, Unlimited.
Maximum Calls
Limit of calls to or from the cloud in this location.
Table continues…

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Location

Field Description
Internal Maximum Default = Unlimited. Range = 1 - 99, Unlimited.
Calls
Limit of calls to or from other configured locations in this location.
Parent Location Default = None.
for CAC
The options are:
• None The default setting.
• Cloud The parent location is an internet address external to the IP Office network.
When set to Cloud, the Call Admission Control (CAC) settings are disabled. Calls to
this location from other configured locations are counted as external, yet no CAC limits
are applied to the location itself.

Time Settings
For extensions, the display of location based time is only supported on 1100, 1200, 1600, 9600
and J100 Series phones plus D100, E129 and B179 telephones.
Field Description
Time Zone Default = Same as System
Select a time zone from the list.
• If set to Same as System, then the time zone configured for the system is used:
- For IP500 V2 systems, the time zone is set through the time settings on the System >
System menu.
- For Linux based servers, the time zone is set through the server’s Platform View
menus.
• When set to a specific timezone, the settings below are also usable to further adjust the
time.

Field Description
Local Time Offset Default = Based on the selected locale and time zone. See Avaya IP Office Locale
from UTC Settings.
This setting is used to set the local time difference from the UTC time value provided by
SNTP. For example, if the system is 5 hours behind UTC, configured this field as -05:00.
• You can adjust the offset in 15 minute increments.
Use this offset for the standard (non-daylight savings time) time. To apply an additional
offset for daylight saving time periods, using the settings below.
Automatic DST Default = Based on the selected locale and time zone. See Avaya IP Office Locale
Settings.
When enabled, the system automatically corrects for daylight saving time (DST) changes
using the settings below.
Table continues…

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 Address

Field Description
Clock Forward/ Default = Based on the selected locale and time zone. See Avaya IP Office Locale
Back Settings Settings.
This field displays entries for when the IP Office should apply and remove a daylight
saving time offset in addition to the Local Time Offset from UTC.
You can configure up to 10 entries (20 for IP Office R11.1.3.2 and higher).
• To edit an entry, select it and then click Edit.
• To delete an entry, select it and click Delete.
• In order to add a new entry you may need to delete an existing entry. The option Add
New Entry then appears at the bottom of the list.
Each entry has the following settings:

Field Description
DST Offset The number of hours to shift the local time for DST.
Clock Select Clock Forward to see and edit when the clock will move
Forward/Back forward to start daylight saving.
Select Clock Back to see and edit when the clock will move
backward to end daylight saving.
Local Time To The time of day to move the clock forward to start daylight saving.
Go Forward

Local Time To The time of day to move the clock backward to end daylight saving.
Go Back
Date for Clock The date for moving the clock forwards or backwards. Select the date
Forward/Back by double-clicking on it in the calendar.

Related links
Location on page 598

Address
Navigation: Location | Address
This information is used for SIP lines to an E911 service supporting RFC 4119 and RFC 5139. On
emergency calls, the address information is included in the INVITE message.
To use the information, the Line | SIP Line | Advanced | Send Location Info settings must be
enabled.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Location

Field Description Example

Country The country is identified by the two letter ISO 3166 code. US
Code
A1 National subdivisions (state, region, province, New York
prefecture).
A2 County, parish, gun (JP), district (IN). King’s County
A3 City, township, shi (JP). New York
A4 City division, borough, city district, ward, chou (JP). Manhattan
A5 Neighborhood, block. Morningside Heights
A6 Street. Broadway
RD Primary road or street Broadway
RDSEC Trailing street suffix. SW
RDBR Road branch. Lane 7
RDSUBBR Road sub-branch. Alley 8
PRD Leading street direction. N
POD Trailing street suffix. NE
STS Street suffix. Avenue, Platz, Street
PRM Road pre-modifier. Old
POM Road post-modifier. Extended
HNO House number, numeric part only. 123
HNS House number suffix. A, 1/2
LMK Landmark or vanity address. Low Library
BLD Building (structure). Hope Theatre
LOC Additional location information. Room 543
PLC Place type. Office
FLR Floor. 5
UNIT Unit (apartment, suite). 12a
ROOM Room. 450F
SEAT Seat (desk, cubicle, workstation). WS 181
NAM Name (residence, business, or office occupant). Joe’s Barbershop
ADDCODE Additional Code 13203000003
PCN Postal community name. Leonia
PC Postal code. 10027-0401
POBOX Post office box (P.O. box) U40

Related links
Location on page 598

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Part 6: Configuring IP Office

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Chapter 51: Subscriptions

Subscriptions are monthly paid entitlements. They can be divided into two main groups;
• per-user per-month user subscriptions
• per-month application subscriptions for selected applications.
In practice, subscriptions are purchased for a specific duration. For example; 6-months, 1-year,
3-years.
During operation:
• If connection to the subscription server is lost, the IP Office system continues running with the
existing subscription entitlements it has already received for 30-days.
• If when connected, any subscription expires, the feature or features associated with the expired
subscriptions cease operation immediately.
- The person responsible for ordering subscriptions must ensure that they are aware of
subscription expiry dates. They must renew subscriptions in a timely manner, including time
for renewal orders to be processed.
Related links
Ordering Subscriptions on page 604
Trial Mode on page 605
User Subscriptions on page 605
Application Subscriptions on page 606
Customer Operations Manager (COM) on page 607
Subscription Connection Operation on page 608
Subscription Network Requirements on page 609
Subscription Mode Ports on page 610
Migrating Existing IP Office Systems to Subscription Mode on page 611

Ordering Subscriptions
Subscription for an IP Office subscription mode system are ordered from the Avaya Channel
Marketplace. The subscriptions are ordered against the PLDS ID of the IP Office system.

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 Trial Mode

After ordering the subscriptions, details of the customer number and address of the subscription
server are supplied in an email. Those details are required during the initial system configuration.
• The person responsible for ordering subscriptions must ensure that they are aware of
subscription expiry dates. They must renew subscriptions in a timely manner, including time
for renewal orders to be processed.
Related links
Subscriptions on page 604

Trial Mode
When ordering an IP Office subscription system through the Avaya Channel Marketplace, trial
mode can be selected. Trial mode enables the IP Office to operate for up to 30-days using free
subscriptions.
• The trial mode IP Office system indicates that it is in 30-day subscription error mode in
applications such as the System Status Application and through system alarms.
• Before the 30-day trial period ends, the subscriber can return to Avaya Channel Marketplace
and request a conversion to paid-subscriptions mode.

Important:
- To avoid any interruptions to customer telephony services, you must request the
change to paid-subscriptions before the end of the 30-day trial period. That request
must include allowance for sufficient working time to implement the request.
Related links
Subscriptions on page 604

User Subscriptions
Each user on the system requires a subscription. All subscribed users are then able to
use an the system's telephone extension (analog, digital or IP) and voicemail features. The
following user subscriptions can be ordered: Telephony User, Telephony Plus User and Unified
Communications User. The subscriptions are applied to individual users through their User
Profile setting.
Feature Subscription Mode
Telephony User Telephony Plus User Unified
Communications User
one-X Portal Services – –
Table continues…

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Subscriptions

Feature Subscription Mode

Telephony User Telephony Plus User Unified
Communications User
Telecommuter options – – ✓
UMS Web Services – –
TTS for Email Reading – –
Remote Worker ✓
Avaya Workplace Client – [1]

WebRTC – –
Mobility Features – –

• By default, users on a new or defaulted system are configured a Telephony User users.
• Users without a subscription are shown as Non-licensed User and cannot use any system
features.
• If there are insufficient subscriptions for the number of users configured to a particular profile,
some of those users will not receive any services. On suitable Avaya phones, they display as
logged out and an attempt to log in displays a no license available warning.
1. Only supports Avaya Workplace Client basic mode (telephony and local contacts only).
Related links
Subscriptions on page 604

Application Subscriptions
The following application subscriptions can be ordered for a IP Office subscription system:
Subscription Description
Receptionist This subscription is used to enable the IP Office SoftConsole application to answer and
Console redirect calls. The number of subscriptions allows the matching number of users to be
configured as IP Office SoftConsole users. Those users still require a user subscriptions
for their telephone connection (IP Office SoftConsole is not a softphone).
Avaya Call This subscription enables support for the Avaya Call Reporter application, hosted on a
Reporter separate server.
Avaya Contact This subscription enables support the Avaya Contact Center Select (ACCS) service
Center Select hosted on a separate server.
Table continues…

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 Customer Operations Manager (COM)

Subscription Description
Media Manager This subscription enables support for Media Manager. This can either be locally hosted
on an IP Office Application Server or provided centrally by the same cloud-based servers
providing the system's subscriptions. In either case:
• A local Voicemail Pro service running on an IP Office Application Server is used to do
the actual recording.
• The recordings are then collected by the Media Manager service for archiving.
Third-Party CTI This subscription enables support for CTI connections by third-party applications. This
includes DevLink, DevLink3, Third-party TAPI and TAPI WAV.

Related links
Subscriptions on page 604

Customer Operations Manager (COM)

IP Office subscription services are a set of cloud-based services provided by Avaya to support
IP Office subscription systems. A separate set of these services is provided for each geographic
region to support Avaya business partners and their customer systems in that region.
The key service is Customer Operations Manager (COM). COM provides:
• Subscriptions to the IP Office systems.
• Displays the status of the IP Office systems and information about current alarms, type of
system, software level.
• Each business partner has an account that allows them to access COM but only see their
own customer's systems. They can create additional COM user accounts and control which
of their customer systems those accounts can see.
• Avaya have access to COM for their support staff in order to manage the COM services and
to assists business partners when required.
• COM can provide the files used to customize various features such as phone background
and screen saver images. This can be configure to provide common files to all the business
partner's systems or individual files to individual end-customer systems.
• COM can act as the file server for firmware files used by Vantage phones and Avaya
Workplace Client.
• For full documentation of COM, refer to the Using Customer Operations Manager for IP
Office Subscription Systems manual.
Additional Support Features
A number of additional support services can be enabled through settings in the IP Office system
configuration.

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Subscriptions

Feature Description
Remote Backup/ Subscription systems can automatically upload daily backups to the cloud. In addition,
Restore COM operators can perform both manual backups and restores operation
Remote Upgrade Avaya provide COM with updated IP Office software images. COM operators can use
these to perform immediate or scheduled system upgrades.
Log File Collection Subscriptions systems can automatically upload all available log files to the cloud each
day.
Centralized Administrator connections for IP Office Web Manager, SysMonitor and System Status
Management Application can be routed through COM to the customer's IP Office systems. The
connects use the TLS tunnel used for the subscriptions.
Remote Access Connections for HTTPS and SSH/SFTP connection can also be routed through COM to
the customer IP Office systems. The connects use the TLS tunnel used for subscription.
Co-located Servers When remote access is enabled, access to other servers and services on the same
network as the customer IP Office system can be enabled. That includes access to
non-IP Office servers and services subject to their own authentication.

Related links
Subscriptions on page 604

Subscription Connection Operation

The connection between the IP Office and COM operates are follows:
Outgoing Connection
For the connection from the IP Office to COM:
• The destination is a single static IP address resolved by DNS of the subscription server
address entered during the system's initial configuration.
• The IP Office alternates between TCP ports 443 and 8443 until successful.
• The link uses the HTTP 'WebSocket' protocol and TLS 1.2 with mutual authentication.
• The link carries a regular heartbeat, subscription information and basic details of the IP Office
system (type of servers and software version).
• All other traffic on the link is controlled by the IP Office system settings; there are no access
controls elsewhere.
• If the link is interrupted, the IP Office system goes into a 30-day error mode with daily alarms.
- If connection to the subscription server is lost, the IP Office system continues running with
the existing subscription entitlements it has received for 30-days.
• During the error mode period, all operations and features are unaffected. The system
outputs daily alarms in the system logs.
• Successful reconnection clears the alarms and error mode.
• If the 30-day error mode period expires, all subscription features and telephony are
deactivated.

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 Subscription Network Requirements

- If when connected, any subscriptions expire, the feature or features associated with the
expired subscriptions cease operation immediately.
- • The person responsible for ordering subscriptions must ensure that they are aware of
subscription expiry dates. They must renew subscriptions in a timely manner, including
time for renewal orders to be processed.
Incoming Connection
All incoming traffic from COM is routed to the IP Office through the existing subscription
connection established above. It should not require any additional configuration on the customer
network if the system has successfully obtained it subscriptions.
Related links
Subscriptions on page 604

Subscription Network Requirements

In order to obtain its subscriptions and to be remotely monitored and managed through COM , the
IP Office systems requires the following:
Feature Description
Subscription Details of the customer ID and subscription server address are provided by email. Those
details details are entered during the system's initial configuration.
• For an IP500 V2 SCN, each IP500 V2 requires a License Server Link.
• For a Server Edition deployment, only the Primary server has a License Server Link.
Internet access The system needs to be able to access the external internet. This is normally achieved
during initial configuration of the system by entering the default gateway address of the
outgoing router on the customer network.
• That value is used to configure an default IP route in the system configuration with the
following settings:

IP Route Setting Value

IP Address 0.0.0.0
IP Mask 0.0.0.0
Gateway IP The address of the external network router on the customer
Address network
Destination The IP Office LAN interface (LAN1 or LAN2) which is connected
to the customer network.

• Maximum round trip delay 200ms.

• Minimum connection bandwidth 128kbits/s.
• If the customer firewall or router controls the ports used for outgoing internet access,
ensure that outgoing HTTPS traffic on TCP ports 8443 and 443 are allowed.
Table continues…

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Subscriptions

Feature Description
DNS Service The address of the customer's DNS server or service. If the customer does not have a
specific DNS service, then use 8.8.8.8.
If the customer has their own DNS server:
• Ensure that it is configured to allow external access to addresses in the avaya-
sub.com domain. That domain is used to the COM servers that support subscription
systems in various geographic regions. For example: admin.uk1.avaya-sub.com.
• Ensure that it is also configured to allow external access to storage.googleapis.com.
This address is used for subscription features that require access to file storage.
Time source Subscriptions requires an accurate time source. The recommendation is to use the
Google time service at time.google.com. The system's time zone should also be set
correctly.
COMAdmin The connection from the system to COM uses the security settings of the COMAdmin
Security User service user account in the IP Office system's security settings. This account is created
by default on new and default systems.

Related links
Subscriptions on page 604

Subscription Mode Ports

The following schematic shows the ports used for connections to and from the subscription service
running on COM.

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 Migrating Existing IP Office Systems to Subscription Mode

Related links
Subscriptions on page 604

Migrating Existing IP Office Systems to Subscription Mode

The process for migrating an existing IP Office Essential Edition or Preferred Edition system to
IP Office system is can be performed by rerunning the initial configuration menu. The assumed
mapping of existing user profiles to their subscription equivalents is as follows:
Essential/Preferred Edition Mode Subscription Mode
Non-Licensed User Non-Licensed User
Basic User Telephony User
Mobile User
Office Worker UC User
Power User

Related links
Subscriptions on page 604

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Chapter 52: General System Configuration

This section covers various aspects of IP Office system configuration.

Related links
Centralized System Directory on page 612
Advice of Charge on page 616
Using Locations on page 617
Caller Display on page 617
Parking Calls on page 618
Automatic Intercom Calls on page 619
Wide Band Audio Support on page 620
Media Connection Preservation on page 621
Configuring IP Routes on page 622
Creating a Virtual WAN Port on page 624

Centralized System Directory

Directory services can be used to import directory records (names and numbers) from external
sources. These sets of records are regularly re-imported.
Directory records can come from the following sources:
• LDAP Import: The system can import LDAP records for use within directories shown by
user phones and applications. LDAP import is configured through the System | Directory
Services | LDAP form. You can use LDAP Version 2 and 3.
• HTTP Import : Systems are able to import the directory records from another system using
HTTP. HTTP import is configured through the System | Directory Services | HTTP form
by specifying an IP address or multi-site network connection. The records imported can be
any or all of the following record types held by the system from which the records are being
imported: LDAP imported records, HTTP imported records, configuration records.
• System Directory Records (Configuration records): Records can be entered directly into
the system configuration through the Directory | Directory Entry form. System directory
records override matching LDAP/HTTP imported records.

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 Centralized System Directory

Users with system phone rights (see System Phone Features on page 728) and a phones
with a CONTACTS button can add, delete and edit the system directory records of the
system on which they are logged in. They cannot edit LDAP or HTTP imported records.

Server Edition Directory Operation

For a Server Edition network, these settings can only be configured at the network level and
they are stored in the configuration of the Primary Server. All other systems in the network are
configured to share the directory settings of the Primary Server through the settings at System |
Directory Services | HTTP.
Directory Record Capacity
The directory capacity depends on the type of system. The figures below are applicable for
Release 10.0.
System Number of Directory Records Total Number
Configuration LDAP Import HTTP Import of Directory
Records
Standalone IP500 V2 2,500 10,000 10,000 10,000
Systems
Server Edition Primary Server 10,000 10,000 10,000 10,000
Secondary – – 10,000 10,000
Server
Expansion – – 10,000 10,000
System (L)
Expansion – – 10,000 10,000
System (V2)

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General System Configuration

Directory Dialing
Directory numbers and names are displayed by user applications such as SoftConsole. The
method by which these directories are searched and used depends on the application. Refer to
the appropriate user guide.
Directory entries used for dialing can contain () and — characters in the number. Those characters
are ignored in the dialled output. Directory entries containing ? in the number (used for directory
name matching) are not included in the directory for dialing.
Directory names are also viewable through the Dir or Contacts function on many Avaya phones.
They allow the user to select the name in order to dial its associated number.
The directory function groups directory records shown to the phone user into several categories,
for example; system, personal, users and groups. Depending on the phone or application, the
user may be able to select the category currently displayed. In some scenarios, the categories
displayed may be limited to those supported for the action being performed by the user. The
typical categories are:
• External: Directory records from the system configuration. This includes HTTP and LDAP
imported records.
• Groups: Groups on the system. If the system is in a multi-site network, it will also include
groups on other systems in the network.
• Users or Index: Users on the system. If the system is in a multi-site network it will also
include users on other systems in the network.
• Personal: Available on 1400, 1600, 9500, 9600 and J100 Series phones. These are the
user's personal directory records stored within the system configuration.
On phones that support Dir or Contacts, the user can filter the currently displayed set of directory
names by dialing on their keypad. Additional dialing applies a progressive filter. For example, if the
user presses the 5 key (JKL), only names with some part beginning with J, K or L remain listed.
If the user then presses the 2 key (ABC), only names with some part beginning with JA, JB, JC,
KA, etc. remain listed. As the users presses more keys on their phone, the number of remaining
matches reduces.
By default the letter matching is performed simultaneously against all parts of the directory name,
ie. first, middle and last name. However, this behavior can be modified for all users using a
NoUser source number.
Speed Dialing
On M-Series and T-Series phones, a Speed Dial button or dialing Feature 0 can be used to
access personal directory records using the record’s index number.
• Personal: Dial Feature 0 followed by * and the 2-digit index number in the range 01 to 99.
• System: Dial Feature 0 followed by 3-digit index number in the range 001 to 999.
• The Speed Dial short code feature can also be used to access a directory speed dial using
its index number from any type of phone.
Caller Directory Name Matching
Directory records are also used to associate a name with the dialled number on outgoing calls or
the received CLI on incoming calls. When name matching is being done, a match in the user's

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 Centralized System Directory

personal directory overrides any match in the system directory. Note that some user applications
also have their own user directory.
• The ( ) and — characters are not used for directory name matching. Directory entries with
those characters are ignored for name matching.
• A ? character can be used to match any digit or digits. For example 91?3 will match 9123.
Typically a single ? is used at the end of a known dialing string such as an area code.
• The best match is used, determined by the highest number of matched digits.
• There is no minimum number of matches. For example, a directory entry of 9/External can be
used to match any external call unless it has a better match.
Other Name Sources
• SoftConsole has its own directories which are also used for name matching. Matches in the
application directory can lead to the application displaying a different name from that shown
on the phone.
• Name matching is not performed when a name is supplied with the incoming call, for
example QSIG trunks. On SIP trunks the use of the name matching or the name supplied by
the trunk can be selected using the Default Name Priority setting (System | Telephony |
Telephony). This setting can also be adjusted on individual SIP lines to override the system
setting.
• Directory name matching is not supported for DECT handsets. For information on directory
integration, see IP Office DECT R4 Installation.
Imported Records
Imported directory records are temporary until the next import refresh. They are not added to the
system's configuration. They cannot be viewed or edited using Manager or edited by a system
phone user. The temporary records are lost if the system is restarted. However the system will
request a new set of imported directory records after a system restart. The temporary records
are lost if a configuration containing Directory changes is merged. The system will then import a
new set of temporary records without waiting for the Resync Interval. If an configuration record is
edited by a system phone user (see System Phone Features on page 728) to match the name or
number of a temporary record, the matching temporary record is discarded.
Importation Rules:
When a set of directory records is imported by HTTP or LDAP, the following rules are applied to
the new records:
• Imported records with a blank name or number are discarded.
• Imported records that match the name or number of any existing record are discarded.
• When the total number of directory records has reached the system limit, any further
imported records are discarded.
For capacity information, see the description for the Directory tab.
Related links
General System Configuration on page 612

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General System Configuration

Advice of Charge
The system supports advice of charge (AOC) on outgoing calls to ISDN exchanges that provide
AOC information. It supports AOC during a call (AOC-D) and at the end of a call (AOC-E). This
information is included in the SMDR output.
AOC is only supported on outgoing ISDN exchange calls. It is not supported on incoming calls,
reverse charge calls, QSIG and non-ISDN calls. Provision of AOC signalling will need to be
requested from the ISDN service provider and a charge may be made for this service.
The user who makes an outgoing call is assigned its charges whilst they are connected to the call,
have the call on hold or have the call parked.
• If AOC-D is not available, then all charges indicated by AOC-E are assigned to the user who
dialed the call.
• If AOC-D is available:
- If the call is transferred (using transfer, unpark or any other method) to another user, any
call charges from the time of transfer are assigned to the new user.
- If the call is manually transferred off-switch, the call charges remain assigned to the user
who transferred the call.
- If the call is automatically forwarded off switch, subsequent call charges are assigned to
the forwarding user.
- AOC-D information will only be shown whilst the call is connected. It will not be shown
when a call is parked or held.
- Call charges are updated every 5 seconds.
For conference calls all call charges for any outgoing calls that are included in the conference
are assigned to the user who setup the conference, even if that user has subsequently left the
conference.
Enabling AOC Operation
1. Set the System Currency The Default Currency (System | Telephony | Telephony) setting
is by default set to match the system locale. Note that changing the currency clears all call
costs stored by the system except those already logged through SMDR.
2. Set the Call Cost per Charge Unit for the Line AOC can be indicated by the ISDN
exchange in charge units rather than actual cost. The cost per unit is determined by the
system using the Call Cost per Charge Unit setting which needs to be set for each line.
The values are 1/10,000th of a currency unit. For example if the call cost per unit is £1.07,
a value of 10700 should be set on the line.
3. Applying a Call Cost Markup It may be a requirement that the cost applied to a user's
calls has a mark-up (multiplier) applied to it. This can be done using the Call Cost Markup
(User | Telephony | Call Settings) setting. The field is in units of 1/100th, for example an
entry of 100 is a markup factor of 1.
Related links
General System Configuration on page 612

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 Using Locations

Using Locations
Locations are used to apply an number of common settings to lines and extensions that are in the
same physical location. For example:
• Apply restrictions to the number of simultaneous calls on internal trunks between different IP
Office systems. See Configuring Call Admission Control on page 709.
• Set the outgoing ARS that should be used when an extension associated with the location
makes an emergency call. The aim being to ensure that emergency calls use trunks that
match their physical location or using a caller ID number registered to the location. See
Configuration for Emergency Calls on page 652.
For SIP trunks, emergency calls can include sending the address information configured for
the dialing extension's location.
• Apply location specific time offset settings to the time display on phones in the location.
Related links
General System Configuration on page 612

Caller Display
Caller display displays details about the caller and the number that they called. On internal
calls, the system provides this information. On external calls it uses the Incoming Caller Line
Identification (ICLID) received with the call. The number is also passed to system applications and
can be used for features such as call logging, missed calls and to make return calls.
Analog extension can be configured for caller display via the system configuration (Extension |
Extn | Caller Display Type).
Adding the Dialing Prefix Some systems are configured to require a dialing prefix in front of
external numbers when making outgoing calls. When this is the case, the same prefix must be
added to the ICLID received to ensure that it can be used for return calls. The prefix to add is
specified through the Prefix field of each line.
Directory Name Matching The system configuration contains a directory of names and numbers.
If the ICLID of an incoming call matches a number in the directory, the directory name is
associated with that call and displayed on suitable receiving phones.
Applications such as SoftConsole also have directories that can be used for name matching.
If a match occurs, it overrides the system directory name match for the name shown by that
application.
Extended Length Name Display
In some locales, it may be desirable to change the way names are displayed on phones in order to
maximize the space available for the called or calling name. There are two hidden controls which
can be used to alter the way the system displays calling and called information.

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General System Configuration

These controls are activated by entering special strings on the Source Numbers tab of the NoUser
user. These strings are:
LONGER_NAMES This setting has the following effects:
• On DS phones, the call status display is moved to allow the called/calling name to occupy the
complete upper line and if necessary wrap-over to the second line.
• For all phone types:
• On incoming calls, only the calling name is displayed. This applies even to calls forwarded
from another user.
• On outgoing calls, only the called name is displayed.
HIDE_CALL_STATE This settings hides the display of the call state, for example CONN when
a call is connected. This option is typically used in conjunction with LONGER_NAMES above to
provide additional space for name display.
Related links
General System Configuration on page 612

Parking Calls
Parking a call is an alternative to holding a call. A call parked on the system can be retrieved by
any other user if they know the system park slot number used to park the call. When the call is
retrieved, the action is known as Unpark Call or Ride Call. While parked, the caller hears music on
hold if available.
Each parked call requires a park slot number. Attempting to park a call into a park slot that is
already occupied causes an intercept tone to be played. Most park functions can be used either
with or without a specified park slot number. When parking a call without specifying the park slot
number, the system automatically assigns a number based on the extension number of the person
parking the call plus an extra digit 0 to 9. For example if 220 parks a call, it is assigned the park
slot number 2200, if they park another call while the first is still parked, the next parked call is
given the park slot number 2201 and so on.
Park slot IDs can be up to 9 digits in length. Names can also be used for application park slots.
The Park Timeout setting in the system configuration (System | Telephony | Telephony | Park
Timeout) controls how long a call can be left parked before it recalls to the user that parked it.
The default time out is 5 minutes. Note that the recall only occurs if the user is idle has no other
connected call.
There are several different methods by which calls can be parked and unparked. These are:
Using Short Codes

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 Automatic Intercom Calls

The short code features, Call Park and Unpark Call, can be used to create short codes to park and
unpark calls respectively. The default short codes that use these features are:
• *37*N# - Parks a call in park slot number N.
• *38*N# - Unparks the call in park slot number N.
Using the SoftConsole Application
The SoftConsole application supports park buttons. SoftConsole provides 16 park slot buttons
numbered 1 to 16 by default.
The park slot number for each button can be changed if required. Clicking on the buttons allows
the user to park or unpark calls in the park slot associated with each button. In addition, when a
call is parked in one of those slots by another user, the application user can see details of the call
and can unpark it at their extension.
Using Programmable Buttons
The Call Park feature can be used to park and unpark calls. If configured with a specified park
slot number, the button can be used to park a call in that slot, unpark a call from that slot and will
indicate when another user has parked a call in that slot. If configured without a number, it can be
used to park up to 10 calls and to unpark any of those calls.
Phone Defaults
Some telephones support facilities to park and unpark calls through their display menu options
(refer to the appropriate telephone user guide). In this case parked calls are automatically put into
park slots matching the extension number.
Related links
General System Configuration on page 612

Automatic Intercom Calls

Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
Making Automatic Intercom Calls
The following programmable button functions can be used to make automatic intercom calls:
• Automatic Intercom
• Dial Direct
• Dial Intercom
The following short code function can be used to make automatic intercom calls:
Dial Direct

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General System Configuration

On M-Series and T-Series phones, the code Feature 66 followed by the extension number can be
used to make a direct voice (automatic intercom) call.
Deny automatic intercom calls
When enabled, any automatic intercom calls to the user's extension are automatically turned into
normal calls.
Deny automatic intercom calls can be configured per user on the User | Telephony | Supervisor
Settings tab. Deny automatic intercom call can also be enabled using the Auto Intercom Deny On
short code or the Auto Intercom Deny button action.
Related links
General System Configuration on page 612

Wide Band Audio Support

IP Office systems support the G.722 64K codec for wide band audio. G.722 can be used with
H.323 and SIP trunks. If can also be used with some SIP and H.323 IP telephones (see below).
G.722 uses a higher speech sample rate (16KHz) than is used by most other audio codecs
(8KHz).
G.722 is only supported by systems that are using IP500 VCM, IP500 VCM V2 and or IP500
Combination cards.
Avaya Phone Support
Use of G.722 is supported by the following Avaya phones on a IP Office system: 1100/1200
Series, 9600 Series, J100 Series, B179, B199.
Using the G.722 Codec
The G.722 codec is not available for use by default. If the codec is to be used, it must first be
selected in the system's Available Codecs list (System | Codecs). The codec can then be used
in the system's default codec preference list and or in the individual codec preferences of IP lines
and extensions.
The method of codec selection for specific phones will depend on the phone type. Refer to the
appropriate installation manual.
Conferencing
Where devices using G.722 are in a system conference, the system can attempt to ensures that
the speech between devices using G.722 remains wide-band even if there are also narrow-band
audio devices in the same conference. This is done if the system's High Quality Conferencing
option is enabled (System | Telephony | Telephony).
Known Limitations
The following limitations apply to G.722 wide band audio operation:
• Call recording uses G.711.
• Page calls only use G.722 when all devices being paged can use G.722.

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 Media Connection Preservation

• Fax is not supported in G.722, use G.711 or T38.

• Soft tones provided by the system use G.711.
• A maximum of 15 G.722 devices receiving wide-band audio are supported in conferences.
Related links
General System Configuration on page 612

Media Connection Preservation

Media Connection Preservation maintains calls that experience end-to-end signalling loss or
refresh failures but that still have an active media path.
IP Phones:
With IP Office 9.1 and higher, the following Avaya IP phones attempt to maintain calls when the
signal from the host IP Office is lost.
• 9608
• 9611
• 9621
• 9641
• J100 Series
When preserving a call, the phone does not attempt to reregister with their call server or attempt
to failover to a standby call server, until the call has ended. Softkey call actions and feature menus
do not work during this time due to the loss of signalling path. The phone display is not updated
and the only permitted action is to terminate the call.
IP Office:
When enabled for a particular IP endpoint type that supports Media Connection Preservation, the
call is put into a Preserved state and a Preservation Interval timer is started for that call at the
point the signalling loss is detected. The maximum duration of a preserved call on IP Office is two
hours. Once put into the Preserved state, a call can only transition to the Terminated state. Call
restoration is not supported.
Only the following call types are preserved:
• Connected active calls
• Two party calls where the other end is a phone, trunk, or voicemail
• Conference calls
• Calls on hold and calls to hunt groups are not preserved.
Phone Display:
When a call is in a preserved state but the phone's local signalling connection with its host IP
Office is still present, the phone call state display is prefixed with a warning icon. Hold, transfer,
and conference actions are not available.

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General System Configuration

System Configuration
When enabled on System | Telephony | Telephony, Media Connection Preservation is applied
at a system level to SCN trunks and Avaya H.323 phones that support connection preservation.
All systems in a Small Community Network (SCN) must be enabled for end to end connection
preservation to be supported.
When enabled on Line | SIP Line | SIP Advanced, Media Connection Preservation is applied
to the SIP trunk. The value of connection preservation on public SIP trunks is limited. Media
Connection Preservation on public SIP trunks is not supported until tested with a specific service
provider. Media Connection Preservation is disabled by default for SIP trunks.
When enabled on Line | SM Line | Session Manager, Media Connection Preservation is applied
to Enterprise Branch deployments. Media Connection Preservation preserves only the media and
not the call signaling on the SM Line. Media Connection Preservation does not include support for
the Avaya Aura Session Manager Call Preservation feature.
Related links
General System Configuration on page 612

Configuring IP Routes
The system acts as the default gateway for its DHCP clients. It can also be specified as the default
gateway for devices with static IP addresses on the same subnet as the system. When devices
want to send data to IP addresses on different subnets, they will send that data to the system as
their default gateway for onward routing.
The IP Route table is used by the system to determine where data traffic should be forwarded.
This is done by matching details of the destination IP address to IP Route records and then using
the Destination specified by the matching IP route. These are referred to as 'static routes'.
Automatic Routing (RIP): The system can support RIP (Routing Information Protocol) on LAN1
and or LAN2. This is a method through which the system can automatically learn routes for data
traffic from other routers that also support matching RIP options, see RIP. These are referred to as
'dynamic routes'. This option is not supported on Linux based servers.
Dynamic versus Static Routes: By default, static routes entered into the system override any
dynamic routes it learns by the use of RIP. This behavior is controlled by the Favor RIP Routes
over static routes option on the System | System tab.
Static IP Route Destinations: The system allows the following to be used as the destinations for
IP routes:
• LAN1 Direct the traffic to the system's LAN1.
• LAN2 Traffic can be directed to LAN2.
• Service Traffic can be directed to a service. The service defines the details necessary to
connect to a remote data service.
• Tunnel Traffic can be directed to an IPSec or L2TP tunnel.

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 Configuring IP Routes

Default Route: The system provides two methods of defining a default route for IP traffic that
does not match any other specified routes. Use either of the following methods:
• Default Service Within the settings for services, one service can be set as the Default
Route (Service | Service).
• Default IP Route Create an IP Route record with a blank IP Address and blank IP Mask set
to the required destination for default traffic.
RIP Dynamic Routing common
Routing Information Protocol (RIP) is a protocol which allows routers within a network to exchange
routes of which they are aware approximately every 30 seconds. Through this process, each
router adds devices and routes in the network to its routing table.
Each router to router link is called a 'hop' and routes of up to 15 hops are created in the routing
tables. When more than one route to a destination exists, the route with the lowest metric (number
of hops) is added to the routing table.
When an existing route becomes unavailable, after 5 minutes it is marked as requiring 'infinite' (16
hops). It is then advertised as such to other routers for the next few updates before being removed
from the routing table. The system also uses 'split horizon' and 'poison reverse'.
RIP is a simple method for automatic route sharing and updating within small homogeneous
networks. It allows alternate routes to be advertised when an existing route fails. Within a large
network the exchange of routing information every 30 seconds can create excessive traffic. In
addition the routing table held by each system is limited to 100 routes (including static and internal
routes).
It can be enabled on LAN1, LAN2 and individual services. The normal default is for RIP to be
disabled.
• Listen Only (Passive): The system listens to RIP1 and RIP2 messages and uses these to
update its routing table. However the system does not respond.
• RIP1: The system listens to RIP1 and RIP2 messages. It advertises its own routes in a RIP1
sub-network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): The system listens to RIP1 and RIP2 messages. It
advertises its own routes in a RIP2 sub-network broadcast. This method is compatible with
RIP1 routers.
• RIP2 Multicast: The system listens to RIP1 and RIP2 messages. It advertises its own routes
to the RIP2 multicast address (249.0.0.0). This method is not compatible with RIP1 routers.
Broadcast and multicast routes (those with addresses such as 255.255.255.255 and 224.0.0.0)
are not included in RIP broadcasts. Static routes (those in the IP Route table) take precedence
over a RIP route when the two routes have the same metric.
Related links
General System Configuration on page 612

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General System Configuration

Creating a Virtual WAN Port

Procedure

1. Select WAN Port.

2. Click and select PPP.
3. In the Name field, enter either LINEx.y where:
• LINE must be in uppercase.
• x is the line number. For a PRI/T1 module in Slot A, this will be 1. For a PRI/T1 module
in Slot B, this will be 5.
• y is the lowest numbered channel number to be used by the WAN link minus 1. For
example, if the lowest channel to be used is channel 1 then y = 1 - 1 = 0.
4. In the Speed field, enter the total combined speed of the maximum number of channels
sets in the Service.
In this example, 12 channels x 64000 bits = 76800.

Note:
The maximum number of channels that can be used will be limited by the number of
data channels supported by the system Control Unit and not already in use.
5. In the RAS Name field, select the RAS name created when the new Service of that name
was created.
6. Click OK.
Related links
General System Configuration on page 612

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Chapter 53: On-boarding

On-boarding refers to the configuration of an SSL VPN service in order to enable remote
management services to customers, such as fault management, monitoring, and administration.
You must use the Web Manager client to configure on-boarding.
For full details on how to configure and administer SSL VPN services, refer to Deploying Avaya IP
Office™ Platform SSL VPN Services.
The procedure provided below configures IP Office for Avaya support services. Avaya partners
can also use an SSL VPN to provide support services.
Related links
Configuring an SSL VPN using an on-boarding file on page 625

Configuring an SSL VPN using an on-boarding file

The on-boarding XML file is available from Avaya. It contains the settings required to establish a
secure tunnel between IP Office and an AVG server. When you import the on-boarding XML file, it
applies the settings and installs one or multiple TLS certificates.
When you configure the SSL VPN service on a new system, you must begin by generating an
inventory file of the IP Office system. When you register your IP Office system, the inventory file
that you generated is uploaded to the GRT and the inventory data is populated in the Avaya
Customer Support (ACS) database. After you enable remote support, you can download the XML
on-boarding file from the GRT web site and upload it into your IP Office system.
The on-boarding process configures:
• SSL VPN service configuration
• short codes for enabling and disabling the SSL VPN service
• SNMP alarm traps
• one or more TLS certificates in the IP Office trusted certificate store
Perform this procedure using the Avaya IP Office Web Manager client.

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On-boarding

Warning:
The process of 'on-boarding automatically creates an SSL VPN service in the system
configuration when the on-boarding file is uploaded to the system. Care should be taken
not to delete or modify such a service except when advised to by Avaya.
Before you begin
Before you begin, you must have the hardware codes and catalog description of your IP Office
system. For example, “IP OFFICE 500 VERSION 2 CONTROL UNIT TAA” is a hardware code and
catalog description.
Procedure
1. Select Tools > On-boarding.
The On-boarding dialog box displays.
2. If the hardware code for your IP Office system ends with the letters TAA, select the
checkbox next to the prompt Are you using TAA series hardware?
3. Click Get Inventory File to generate an inventory of your IP Office system.
4. Click Register IP Office.
A browser opens and navigates to the GRT web site.
5. Log in to the web site and enter the required data for the IP Office system.
6. Select Remote Support for the IP Office system.
7. Click Download and save the on-boarding file.
8. Browse to the location where you saved the on-boarding file and click Upload.
A message displays to confirm that the on-boarding file has installed successfully.
Related links
On-boarding on page 625

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Chapter 54: Fax Support

Fax on IP500 V2 Systems

IP500 V2 systems can terminate T38 fax calls. For a system with an IP500 VCM, IP500 VCM V2
or IP500 Combo cards, T38 or G.711 can be used for fax transmission. Each fax call uses a VCM
channel unless it is a T38 fax call between compatibly configured call legs. A SIP line or extension
must support Re-Invite.
T38 Fallback can also be specified. On outgoing fax calls, if the called destination does not support
T38, a re-invite is sent for fax transport using G.711.
Configuring Fax on SIP Lines and Extensions:
To configure Fax on SIP Lines and Extensions:
1. On the VoIP page for the line or extension, set Re-Invite Supported to On in order to
enable Fax Transport Support
2. Select a value in the Fax Transport Support field.
Note the following:
• Direct media is supported.
• If Fax Transport Support is set to T38 or T38 Fallback, the T38 Fax page is available. The
T38 Fax page provides detailed T38 configuration options.
Configuring Fax on an IP Office Line:
Within a multi-site network, Fax Transport Support can also be enabled on the IP Office Lines
between the systems. This allows fax calls at one system to be sent to another system.
To configure Fax on an IP Office Line:
1. Set IP Office Line | Line Settings | Networking Level to SCN.
2. Set IP Office Line | VoIP | Fax Transport Support to Fax Relay.
Related links
Server Edition T38 Fax Support on page 628

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Fax Support

Server Edition T38 Fax Support

PSTN
SIP
Provider

IP Office Line
Primary IP500 V2
Server Expansion

SIP Fax Analog

ATA Fax

Fax on Server Edition Linux Servers

IP Office Linux servers cannot terminate T38 fax and therefore, T38 is negotiated end-to-end.
When a SIP ATA fax is connected to an IP Office Linux server, the system directly relays
negotiation between the SIP ATA Fax and the SIP provider.
Configuring Fax on SIP Lines and Extensions:
To configure Fax on SIP Lines and Extensions, on the VoIP page for the SIP line or extension:
1. Set Re-Invite Supported to On in order to enable Fax Transport Support.
2. Select a value in the Fax Transport Support field.
Note the following.
• Direct media is supported.
• The T38 Fax page is not available.
Fax on Server Edition IP500 V2 Expansion Systems
Since an IP500 V2 system can terminate T38 fax, an analog fax can be connected to an IP500
V2 Expansion system. Fax transport is configured on the IP Office Line connecting the IP500 V2
system to the Server Edition network.

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 Server Edition T38 Fax Support

Configuring Fax on an IP Office Line:

To configure Fax on an IP Office Line, on the Line | IP Office Line | VoIP Settings page, select a
value in the Fax Transport Support field. Fax Relay is not supported.
Note the following.
• Direct media is supported.
• The T38 Fax page is not available.
Related links
Fax Support on page 627

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Chapter 55: Paging

The IP Office supports flexible paging to any extensions that support auto-answer and also paging
to external paging devices. However, no paging options are configured by default on a newly
installed IP Office system.
Paging Scenarios
Paging Scenario Paged Device Connects to... Short Code/ Button Feature
Phone to Phone Digital Station and Avaya H.323 Dial Paging
Phones
Simple paging to other system
extensions.
Mixed Paging Analog Extension (Paging Speaker) Dial Paging
Simultaneous paging to phones
and a paging speaker.
Paging Interface Device Analog Extension (IVR Port) Dial Extn
Paging to a paging interface device Analog Trunk Dial
such as a UPAM.

Related links
Paging Capacity on page 630
Phone to Phone Paging on page 631
Paging to an External Paging Device on page 631
Mixed Paging on page 632
Paging Via Voicemail Pro on page 633

Paging Capacity
For full capacity details, refer to Avaya IP Office™ Platform Guidelines: Capacity.

IP Office Type Paging Group Maximum size

Server Edition/Select 512
IP500 V2 64

• Paging groups that include users on a V2 Expansion are limited to 64 members.

• Paging groups that include SRTP endpoints, the maximum size is reduce by 50%.

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 Phone to Phone Paging

Related links
Paging on page 630

Phone to Phone Paging

• Paging is supported from all phone types. A page call can be to a single phone or a group of
phones.
- From analog and non-Avaya phones, use a Dial Paging short code.
- From Avaya feature phones, a programmable button set to Dial Paging can be used.
• Paging is only supported to Avaya phones that support auto answer.
• The page is not heard on phones that are active on another call.
• The page is not heard on phones where the user is set to Do Not Disturb or has Forward
Unconditional active.
• On Avaya phones with a dedicated Conference button, the user can answer a page call by
pressing that button. This turns the page into a normal call with the pager.
Related links
Paging on page 630

Paging to an External Paging Device

Paging Interface Device
Uses a paging interface device such as a UPAM or amplifier with analog trunk/extension interface.
The device can be connected to an analog trunk port or analog extension port.

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Paging

If connected to a trunk port:

• Use a short code set to Dial and the same Line Group ID as the Outgoing Line ID set for
the analog trunk.
If connected to an extension port:
• Set the analog extension as an IVR Port in the system configuration (Extn > Analog >
Equipment Classification).
• Use a short code or programmable button set to use Dial Extn.
Related links
Paging on page 630

Mixed Paging

Uses an amplifier connected to an analog extension port via a 600ohm isolating transformer.
Some amplifiers include an integral transformer. Avaya/Lucent branded amplifiers are designed
for connection to special paging output ports not provided on systems. They are not suitable for
supporting mixed paging.
The transformer and amplifier must be connected when the system is restarted.
If background music is required between pages, the amplifier must support a separate background
music connection and VOX switching.

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 Paging Via Voicemail Pro

The analog extension port is set as a Paging Speaker in the system configuration (Extn | Analog |
Equipment Classification).
Short code/programmable button: Use DialPaging.
Related links
Paging on page 630

Paging Via Voicemail Pro

Voicemail Pro can be used to deliver pre-recorded announcements. This can be useful when the
same announcement is repeated frequently. This method requires the paging port to be an analog
extension.
This method also removes the feedback loop that can occur on some sites as the page is first
recorded and then played.
Example 1
1. In Voicemail Pro, a new Module was added and named Page.

2. A Post Dial action was added to the module. The properties of the Specific tab were set as
shown:

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Paging

3. We then saved and made live the new Voicemail Pro call flow.
4. In Manager we received the system configuration and created a new short code.
• Short Code: *80
• Telephone Number : "Page"
• Feature: VoicemailCollect.
The new system configuration was then merged.
Example 2
This example builds on example 1 by allowing the user to select which message is played from a
menu. In this example the user can press 1, 2 or 3 for different messages. They can also re-record
the message associated with option 3 by pressing #.

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 Paging Via Voicemail Pro

A Play List action was added and in this example set to record pagemsg3.wav. Note that just the
file name was specified as this action saves files relative to the Voicemail Server's WAVS folder.

In the Post Dial action that plays back pagemsg3.wav note that the full file path needs to be used.
In Manager, we then added a short code that triggers the module "Paging" using the
VoicemailCollect feature.
Related links
Paging on page 630

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Chapter 56: System Events

The system supports a number of methods by which events occurring on the system can be
reported. These are in addition to the real-time and historical reports available through the System
Status Application (SSA).
SNMP Reporting
Simple Network Management Protocol (SNMP) allows SNMP clients and servers to exchange
information. SNMP clients are built into devices such as network routers, server PC's, etc. SNMP
servers are typically PC application which receive and/or request SNMP information. The system
SNMP client allows the system to respond to SNMP polling and to send alarm information to SNMP
servers.
In order for an SNMP server application to interact with a system, the MIB files provided with the
Manager installation software must be compiled into the SNMP server's applications database.
Note:
• The process of 'on-boarding' (refer to the Deploying Avaya IP Office™ Platform SSL VPN
Services) may automatically configure SNMP and create a number of SNMP alarm traps.
These will override any existing SNMP configuration settings.
SMTP Email Reporting
The system can send alarms to an SMTP email server. Using SMTP requires details of a valid
SMTP email account user name and password and server address. If SMTP email alarms are
configured but for some reason the system cannot connect with the SMTP server, only the last 10
alarms are stored for sending when connection is successful. Use of SMTP alarms requires the
SMTP server details to be entered in the SMTP tab.
Syslog Reporting
The system can also send alarms to a Syslog server (RFC 3164) without needing to configure an
SNMP server. In addition Syslog output can include audit trail events.
Multiple event destinations can be created, each specifying which events and alarms to include, the
method of reporting to use (SNMP, Syslog or Email) and where to send the events. Up to 2 alarm
destinations can be configured for SNMP, 2 for Syslog and 3 for SMTP email.
Related links
Configuring Alarm Destinations on page 637

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 Configuring Alarm Destinations

Configuring Alarm Destinations

About this task
The Alarms section of the System Events tab displays the currently created alarm traps. It shows
the event destinations and the types of alarms that will trigger the send of event reports. Up to 2
alarm destinations can be configured for SNMP, 2 for Syslog and 3 for SMTP email.
Procedure
1. In the navigation pane, select System.
2. In the details pane, select System Events and then select the Alarms sub-tab.
3. Use the Add, Remove and Edit controls to alter the traps.
4. Click Add or select the alarm to alter and then click Edit.
5. For a new alarm, set the Destination to either Trap (SNMP) or Syslog or Email (SMTP).
Note that once a destination has been saved by clicking OK it cannot be changed to
another sending mode.
6. The remaining details will indicate the required destination information and allow selection
of the alarm events to include.
7. When completed, click OK.
8. Click OKagain.
Related links
System Events on page 636

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Chapter 57: Certificate Management

This section provides an overview of IP Office certificate support and management. For more
comprehensive information, refer to the Avaya IP Office™ Platform Security Guidelines manual.
Related links
Certificate Overview on page 638
Certificate Support on page 643

Certificate Overview
Public key cryptography is one of the ways to maintain a trustworthy networking environment.
A public key certificate (also known as a digital certificate or identity certificate) is an electronic
document used to prove ownership of a public key. The certificate includes information about the
key, information about its owner's identity, and the digital signature of an entity that has verified the
certificate's contents are correct. If the signature is valid, and the person examining the certificate
trusts the signer, then they know they can use that key to communicate with its owner.
The system used to provide public-key encryption and digital signature services is called a public
key infrastructure (PKI). All users of a PKI should have a registered identity which is stored in a
digital format and called an Identity Certificate. Certificate Authorities are the people, processes
and tools that create these digital identities and bind user names to public keys.
There are two types of certificate authorities (CAs), root CAs and intermediate CAs. In order for
a certificate to be trusted and for a secure connection to be established, that certificate must
have been issued by a CA that is included in the trusted certificate store of the device that is
connecting. If the certificate was not issued by a trusted CA, the connecting device then checks to
see if the certificate of the issuing CA was issued by a trusted CA, and so on until either a trusted
CA is found. The trusted certificate store of each device in the PKI must contain the required
certificate chains for validation.

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 Certificate Overview

IP Office Root Certificate Authority

IP Office generates a self-signed certificate. For IP500 V2 systems, a certificate is generated
automatically on the first start up. On Linux systems, a certificate is generated during the ignition
process.
The following entities can act as the certificate authority.
• The Server Edition Primary Server or an Application Server can act as the root certificate
authority for all nodes in the system.
• In Enterprise Branch deployments, the System Manager can act as the root certificate
authority.
• Identity certificates can also be purchased and issued by a third party certificate authority.
Regardless of the method used to provide the IP Office identity, the certificate authority which
signs the IP Office identity certificate must be trusted by all the clients and endpoints that need to
establish a secure connection with IP Office. They must be a part of the PKI. Therefore, the root
CA certificate must be downloaded to client devices and placed in the trusted certificate store. If
there are intermediate CAs in the certificate chain, either the intermediate CAs must be added to
the client device Trusted Certificate Store or the certificate chain must be advertised by IP Office in
the initial TLS exchange.
Certificates and TLS
Telephony signaling like SIP messaging is secured using Transport Layer Security (TLS). TLS
provides communication security using certificates to authenticate the other end of the IP Link.
The message exchange in TLS is aimed at verifying the identity of the communicating parties
and establishing the keys that will be used to encrypt the signaling data between the two parties.

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Certificate Management

Typically, the server sends its identity certificate, either self-signed or signed by the CA, to the
client. The client must have the CA certificate in its trusted certificate store.
IP Office acts as the TLS server in its interactions with SIP telephony clients. This means that the
TLS application on the IP Office must be configured to listen for client connections by enabling
TLS in the SIP Registrar on the LAN1 and LAN2 interfaces.
Note:
• Authentication of the client's certificate by the server is not a requirement. IP Office does
not support client certificate validation for all SIP endpoint types.
• The E.129 phone does not validate the IP Office identity certificate.
Related links
Certificate Management on page 638
Windows Certificate Store on page 640

Windows Certificate Store

The certificate store used by Manager to save and retrieve X509 certificates is the default one
provided by the Windows operating system. The Windows certificate store is relevant to any
application running on Windows that uses certificates for security, either TLS or HTTPS.
Warning:
• Avaya accepts no responsibility for changes made by users to the Windows
operating system. Users are responsible for ensuring that they have read all relevant
documentation and are sufficiently trained for the task being performed.
Windows Certificate Store Organization
By default, certificates are stored in the following structure:

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 Certificate Overview

Each of the sub folders has differing usage. The Certificates - Current User area changes with the
currently logged-in Windows user. The Certificate (Local Computer) area does not change with the
currently logged-in Windows user.
Manager only accesses some of the certificate sub folder:

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Certificate Management

Certificates (Local Manager Use

Computer) Folder
Personal | Certificates Folder searched by Manager 1st for matching certificate to send to the system
when requested. Certificate matched by the subject name contained in File |
Preferences | Security | Certificate offered to the system.
Folder accessed whenever ‘Local Machine certificate store’ used for Security
Settings.
Folder searched by Manager for matching certificate when certificate received from
the system, and File | Preferences | Security | Manager Certificate Checks =
Medium or High.
Trusted Root Folder searched by Manager for matching parent certificates when non-self signed
Certification Authorities certificate received from the system, and File | Preferences | Security | Manager
| Certificates Certificate Checks = Medium or High.

Certificates – Current Manager Use

User Folder
Personal | Certificates Folder searched by Manager 2nd for matching certificate (subject name) to send to
the system when requested. Certificate matched by the subject name contained in
File | Preferences | Security | Certificate offered to the system.
Folder accessed whenever ‘Current User certificate store’ used for Security
Settings.
Folder searched by Manager for matching certificate when certificate received from
IP Office, and File | Preferences | Security | Manager Certificate Checks =
Medium or High.
Trusted Root Folder searched by Manager for matching parent certificates when non-self signed
Certification Authorities certificate received from the system, and File | Preferences | Security | Manager
| Certificates Certificate Checks = Medium or High.
Other People | Folder searched by Manager for matching parent certificates when non-self signed
Certificates certificate received from the system, and File | Preferences | Security | Manager
Certificate Checks = Medium or High.

Windows Certificate Store Import

In order to use certificates – either for security settings or Manager operation – they must be
present in the Windows certificate store. Certificates may be placed in the store by the Certificate
Import Wizard. The Certificate Import Wizard can be used whenever a certificate is viewed.
In order for Manager to subsequently access this certificate the Place all certificate in the
following store option must be selected:
• If the certificate is to subsequently identify the system, the Other People folder should be
used.
• If the certificate is to subsequently identify the Manager, the Personal folder should be used,
and the associated private key saved as well.
Certificate Store Export
Any certificate required outside of the Manager PC must be first saved in the Certificate store,
then exported.

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 Certificate Support

If the certificate is to be used for identity checking (that is, to check the far entity of a link) the
certificate alone is sufficient, and should be saved in PEM or DER format.
If the certificate is to be used for identification that is, to identify the near end of a link) the
certificate and private key is required, and should be saved in PKCS#12 format, along with a
password to access the resultant .pfx file.
Related links
Certificate Overview on page 638

Certificate Support
Related links
Certificate Management on page 638
Certificate File Naming and Format on page 643
Identity Certificate on page 644
Trusted Certificate Store on page 646
Signing Certificate on page 647
Certificate File Import on page 649

Certificate File Naming and Format

DER: Distinguished Encoding Rules (DER) format, which is a binary format used to represent a
certificate. Typically used to describe just one certificate, and cannot include a private key.
There are four main encodings/internal formats for certificate files. Note that these are encodings,
not file naming conventions.
PEM: Privacy Enhanced Mail (PEM) is a Base 64 (i.e. ASCII text) encoding of DER, one
certificate is enclosed between ‘-----BEGIN CERTIFICATE-----‘ and ‘-----END CERTIFICATE-----
‘ statements. Can contain a private key enclosed between ‘-----BEGIN PRIVATE KEY -----‘ and
‘-----END BEGIN PRIVATE KEY -----‘ statements. More than one certificate can be included.
PEM can be identified by viewing the file in a text editor. This is an unsecure format and not
recommended for private key use unless it is protected with a password.
PKCS#12: Public Key Cryptography Standard (PKCS) #12. A secure, binary format, encrypted
with a password. Typically used to describe one certificate, and its associated private key, but can
also include other certificates such as the signing certificate(s). This is the recommended format
for private key use.
PKCS#7: A Base 64 (i.e. ASCII text) encoding defined by RFC 2315, one or more certificates are
enclosed between ‘—–BEGIN PKCS—–‘ & ‘—–END PKCS7—–‘ statements. It can contain only
Certificates & Chain certificates but not the private key. Can be identified by viewing the file in a
text editor.
There are many common filename extensions in use:
• .CRT — Can be DER or PEM. Typical extension used by Unix/Android systems’ public
certificates files in DER format.

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Certificate Management

• .CER — Can be DER or PEM. Typical extension used by Microsoft/Java systems’ public
certificates files in PEM format.
• .PEM — Should only be PEM encoded.
• .DER — Should only be DER encoded.
• .p12 — Should only be in PKCS#12 format. Typical extension used by Unix/Android systems’
identity certificates/private key pair files. Same format as .pfx hence can be simply renamed.
• .pfx — Should only be in PKCS#12 format. Typical extension used by Microsoft systems’
identity certificates/private key pair files. Same format as .p12 hence can be simply renamed.
• .pb7 — Should only be in RFC 2315 format. Typical extension used by Microsoft and Java
systems for certificate chains.
Related links
Certificate Support on page 643

Identity Certificate
Feature Support Notes
Import: Public key Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any other sizes
size are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to be
rejected with an informative error.
Import of certificates with 1024 will be imported after a warning ‘The
certificate public key may not be of sufficient strength. Do you wish to
continue?’
Import: Certificate Yes SHA-1, SHA-256 SHA-384, and SHA-512 hashing algorithms must be
signature algorithm supported. Any other SHA2 algorithms are optional.
Import of certificates with SHA-1 will be imported after a warning ‘The
certificate signature algorithm may not be of sufficient strength. Do you wish
to continue?’
Import of certificates with other algorithms (for example MD5, ECC) to be
rejected with an informative error.
Import: Must have Yes Must be supplied.
private key
Reject and informative error that private key has not been supplied
Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject + informative error if a check fails
Table continues…

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 Certificate Support

Feature Support Notes

Import: Certificate Yes Certificates can be varying sizes
up to 4KB
Import: Formats Yes • PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without password.
This shall be the preferred/default option
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• Pasted from clipboard in PEM format (optional)
NOTE that ONLY PKCS#12 file format is acceptable according to 147434–
030–P1, however we cannot control what format customers receive their
certificates in, hence all should be supported
See section below for certificate file import support
Import: Up to 4 other Yes Only supported where management of TCS also available.
certificates in same
• Any intermediate and root CA certificate included in the PKCS#12 file to be
file
imported into the Trusted Certificate store
• The feature is intended for import of intermediate certificates, but can
include unrelated certificates.
• An informative message to the admin if any have been imported
Import: Certificate Yes Where identity certificate is signed by one or more intermediate CAs, search
chain support TCS for matching certificates and include in identity certificate chain.
View: Certificate Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
Contents
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that certificate has expired
• Warning displayed that certificate is nearing expiry (within 60 days).
View: Private Key No Private key must not be viewable
Export: Formats Yes Private key must not be exportable
Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

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Certificate Management

Related links
Certificate Support on page 643

Trusted Certificate Store

Feature Support Notes
Import: RSA Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any other sizes
1024-4096 key size are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to be
rejected with an informative error.
Import: Optional Yes No private key will actually be imported.
private key
Informative message (neither warning or error) that private key has not been
imported
Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject + descriptive error if a check fails
Import: Certificate Yes Certificates can be varying sizes
up to 4KB
Import: Formats Yes • DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without password.
• Pasted from clipboard in PEM format (optional)
Import: Up to 19 Yes All included certificates, up to 20 total. More than 20 in one file can be
other certificates in optionally supported.
same file
Table continues…

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 Certificate Support

Feature Support Notes

View: TCS Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
Certificate
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that a certificate has expired
• Warning displayed that a certificate is nearing expiry (within 60 days).
Export: Formats Yes Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

Related links
Certificate Support on page 643

Signing Certificate
Feature Support Notes
Import: RSA Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any other sizes
1024-4096 key size are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to be
rejected with an informative error.
Import: Must have Yes Must be supplied.
private key
Reject and informative error that private key has not been supplied
Table continues…

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Certificate Management

Feature Support Notes

Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject and informative error if a check fails
Import: Certificate Yes Certificates can be varying sizes
up to 4KB
Import: Formats Yes • PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without password.
This shall be the preferred/default option
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• Pasted from clipboard in PEM format (optional)
NOTE that ONLY PKCS#12 file format is acceptable according to 147434–
030–P1, however we cannot control what format customers receive their
certificates in, hence all should be supported
Import: Other No Informative warning that other certificates have not been imported
certificates in same
file
View: TCS Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
Certificate
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that certificate has expired
• Warning displayed that certificate is nearing expiry (within 60 days).
Table continues…

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 Certificate Support

Feature Support Notes

Renew existing: Yes Regenerate CA keeping all keys and other contents same except:
• notBefore and notAfter dates
• Serial Number
• Thumbprint (Hash of the certificate)
• ??
Can this be done to imported CAs or just internally generated ones?
Create new: Yes Regenerate CA, including keys
Export: Formats Yes Private key must not be exportable
Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

Related links
Certificate Support on page 643

Certificate File Import

File Content Identity Certificate Trusted Certificate Signing Certificate Notes
Import Command Import Command Import Command
DER
DER: 1 certificate No – attempt Yes – attempt No – attempt
rejected with ‘Invalid accepted with rejected with ‘Invalid
certificate format ‘N certificate(s) certificate format
(DER)’ imported into (DER)’
Trusted Certificate
Store’
DER: Any other No – attempt No – attempt No – attempt
content rejected with ‘Invalid rejected with ‘Invalid rejected with ‘Invalid
content (DER)’ content (DER)’ content (DER)’
PKCS#12
PKCS#12: 1 Yes – attempt No – p12/pfx should Yes – attempt
certificate + private accepted with not be offered for file accepted with
key ‘Certificate import selection ‘Certificate import
successful’ successful’
Certificate/key
imported as ID
certificate
Table continues…

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Certificate Management

File Content Identity Certificate Trusted Certificate Signing Certificate Notes

Import Command Import Command Import Command
PKCS#12: 1 Yes – attempt No – p12/pfx should Yes – attempt At least
certificate + private accepted with not be offered for file accepted with 20 certificates
key, 1 or more other ‘Certificate import selection ‘Certificate import supported in the
certificates successful’ successful’ same file
Certificate/key Certificate/key
imported as ID imported as signing
certificate certificate
Other certificates Other certificates
imported into TCS ignored
with ‘N certificate(s)
imported into
Trusted Certificate
Store’
PKCS#12: Any other No – attempt No – p12/pfx should No – attempt
content rejected with ‘Invalid not be offered for file rejected with ‘Invalid
content (PKCS#12)’ selection content (PKCS#12)’
PEM: 1 Certificate No – attempt Yes – attempt No – attempt Certificate can
rejected with ‘Invalid accepted with rejected with ‘Invalid be encrypted on
certificate format ‘N certificate(s) certificate format unencrypted
(PEM – no private imported into (PEM – no private
key)’ Trusted Certificate key)’
Store’
PEM
PEM: N Certificate No – attempt Yes – attempt No – attempt At least
rejected with ‘Invalid accepted with rejected with ‘Invalid 20 certificates
certificate format ‘N certificate(s) certificate format supported in the
(PEM – no private imported into (PEM – no private same file
key)’ Trusted Certificate key)’
Certificate can
Store’
be encrypted on
unencrypted
PEM: 1 Certificate + Yes – attempt No – attempt Yes – attempt Certificate or Key
private key accepted with rejected with ‘Invalid accepted with can be encrypted on
‘Certificate import certificate format ‘Certificate import unencrypted
successful’ (PEM)’ successful’
Certificate/key Certificate/key
imported as ID imported as signing
certificate certificate
Table continues…

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 Certificate Support

File Content Identity Certificate Trusted Certificate Signing Certificate Notes

Import Command Import Command Import Command
PEM: 1 Certificate Yes – attempt Yes – attempt Yes – attempt Private key must be
+ private key, 1 accepted with accepted with accepted with before or after the
or more other ‘Certificate import ‘N certificate(s) ‘Certificate import first certificate
certificates. successful’ imported into successful’
Certificate or Key
Trusted Certificate
Private key must be Certificate/key Certificate/key can be encrypted on
Store’
before or after the imported as ID imported as signing unencrypted
first certificate certificate. First certificate and certificate
private key ignored
Other certificates Other certificates
imported into TCS ignored
with ‘N certificate(s)
imported into
Trusted Certificate
Store’
PEM: Any other No – attempt No – attempt No – attempt Option to include
content rejected with ‘Invalid rejected with ‘Invalid rejected with ‘Invalid more detail of the
content (PEM)’ content (PEM)’ content (PEM)’ rejection cause.e.g.
‘Cannot detect
Identity Certificate’,
‘Too many private
keys’, ‘Unrecognized
header’ etc.

Related links
Certificate Support on page 643

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Chapter 58: Configuration for Emergency
Calls

This page provides a summary of IP Office emergency call handling. For full details, refer to the IP
Office Emergency Call Configuration manual.
The configuration of every system must contain at least one short code using the Dial Emergency
feature. Dial Emergency overrides all external call barring that may have been applied to the user
whose dialing has been matched to the short code. You must still ensure that no other short code or
extension match occurs that would prevent the dialing of an emergency number being matched to
the short code.
The short code (or codes) can be added as a system short code or as an ARS record short code.
If the Dial Emergency short code is added at the solution level, that short code is automatically
replicated into the configuration of all servers in the network and must be suitable for dialing by
users on all systems. Separate Dial Emergency short codes can be added to the configuration
of an individual system. Those short codes will only be useable by users currently hosted on the
system including users who have hotdesked onto an extension supported by the system.
It is the installers responsibility to ensure that a Dial Emergency short code or codes are useable
by all users. It is also their responsibility to ensure that either:
• the trunks via which the resulting call may be routed are matched to the physical location to
which emergency service should be dispatched
• the outgoing calling line ID number sent with the call matches the physical location from which
the user is dialing.
• If the system uses external dialing prefixes, you should also ensure that the dialing of
emergency numbers with and without the prefix is allowed.
The blocking or rerouting of emergency calls to a intermediate destination other than the emergency
response service may be against local and nation laws.
Hot Desking Users
In addition to the location requirements above, you must also remember that for users who hot desk,
from the networks perspective the user's location is that of the system hosting the extension onto
which the user is currently hotdesked. If that is an IP extension then that location is not necessarily
the same as the physical location of the server.
Emergency call setup
Routing of emergency calls is based on a call resolving to a Dial Emergency short code. Based
on the location value for the extension making the call, routing is performed by the Emergency

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 Emergency Call Indication

ARS form configured for that location. You must ensure that the short codes in the ARS use lines
appropriate for emergency calls from that location.
Configuring emergency call routing
At its simplest, Create a Dial Emergency system short code. Note that the Line Group ID value in
the Dial Emergency short code is overridden if the extension's Locations has an Emergency ARS
defined.
1. Create system short codes for each emergency number used in the system locale. The
short codes should use the Dial Emergency feature. Add short codes for the same numbers
dialed with and without any expected external dialing prefixes.
2. Create an emergency ARS. This should containing shorts codes that take the output of the
system short codes created above and dials them to the external trunks that should be used
for emergency calls from the system.
3. Create a Location for the system and set the Emergency ARS to the ARS created above.
4. Set the location as the system's Location value on the System | System page.
5. For each Extn, set the Location defined above.
6. Test the correct operation of emergency dialing.
7. For networks with multiple systems and locations, create additional emergency ARS entries
and locations as necessary to ensure that emergency calls from any location are sent using
appropriate trunks.
Related links
Emergency Call Indication on page 653
System Alarm Output on page 654

Emergency Call Indication

IP Office R11.1 SP1 added support for a 911 View or Emergency View programmable button.
• A button set to this function indicates to users on the same system when the IP Office has
routed an emergency call out one of its external PSTN trunks.
- The button gives a ring and flashes when there is a connected emergency call in progress.
- The button remains lit when there are details of previous emergency calls in the IP Office
system's emergency call history.
• Pressing the button displays details of currently connected emergency calls (the first 10 such
calls).
• After pressing the button, the History option displays details of any previously connected
emergency calls (the first 30 such calls) and allows deletion of those call details.
• On J189 phones, the details include the location name if the IP Office used a Location
record as part of the emergency call routing.
• All users on the IP Office share the same emergency call history information. Changes to the
emergency call history affect the details shown on all phones on the same system.

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Configuration for Emergency Calls

Related links
Configuration for Emergency Calls on page 652

System Alarm Output

You can configure the IP Office system to generate a system alarm for any call that uses a Dial
Emergency short code. In addition to reporting connected calls, the alarms also report emergency
call attempts that fail for reasons such as no available trunks.
Unlike SMDR call records which the IP Office system generates at the end of a call, the IP Office
generates emergency call system alarms immediately a call matches a Dial Emergency short
code. This is important, as the PSAP emergency operator can stay on the line until the first
responders arrive.
You can configure the IP Office to send system alarms to SNMP, syslog, or email. On-site
notification applications can use the alarm message to offer a variety of features. For example:
• Email/IM/SMS/Pager alerts with escalation and acknowledgments.
• Location maps with additional information. For example; hazardous material warnings.
• Emergency call alert displays for reception/security desks.
• Printing of alerts for physical archiving.
Alarm Information
The IP Office provides the following information in the alarm:
• The location name.
• The number dialed by the caller.
• If connected, the called number and ELIN presented on the call. Otherwise, the reason for
failure.
• The extension's current logged in user, otherwise NoUser. For tandem calls, the Trunk ID.
• The extension details and system ID plus:
- For digital and analog extensions, the physical port details.
- For telecommuter and mobile call control users, the external phone number.
- For IP phones and softphone clients, the MAC and IP Address details.
Related links
Configuration for Emergency Calls on page 652

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Chapter 59: Ring Tones

Ring tones can be defined in the following terms.

Distinctive Ringing - Inside, Outside and Ringback:
A distinctive ring tone can be given for each of the different call types: an internal call, an external
call and a ringback calls (voicemail calls, ringback when free calls, calls returning from park, hold or
transfer).
The distinctive ringing patterns used for most non-analog phones are as follows:
• Internal Call: Repeated single-ring.
• External Call: Repeated double-ring.
• Ringback Call: Two short rings followed by a single ring.
Note:
For non-analog extensions, the ringing pattern used for each call type by the system is not
configurable.

Personalized Ringing:
This term refers to control of the ringing sound through the individual phones. For non-analog
phones, while the distinctive ringing patterns cannot be changed, the ringer sound and tone may be
personalized depending on the phone's own options. Refer to the appropriate telephone user guide.
Analog Phone Ringing Patterns

For analog extensions, the ringing pattern used for each call type can be set using the settings
on System | Telephony | Telephony. The setting for an individual user associated with an analog
extension can be configured using the settings on User | Telephony | Call Settings.
Note that changing the pattern for users associated with fax and modem device extensions may
cause those devices to not recognize and answer calls.
The selectable ringing patterns are:
• RingNormal This pattern varies to match the Locale set in the System | System tab. This is
the default for external calls.
• RingType1: 1s ring, 2s off, etc. This is the default for internal calls.
• RingType2: 0.25s ring, 0.25s off, 0.25s ring, 0.25s off, 0.25s ring, 1.75s off, etc. This is the
default for ringback calls.
• RingType3: 0.4s ring, 0.8s off, …
• RingType4: 2s ring, 4s off, …

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Ring Tones

• RingType5: 2s ring, 2s off, …

• RingType6: 0.945s ring, 4.5s off, …
• RingType7: 0.25s ring, 0.24 off, 0.25 ring, 2.25 off, …
• RingType8: 1s ring, 3s off, …
• RingType9: 1s ring, 4s off, …
• RingType0: Same as RingNormal for the United Kingdom locale.
• Default Ring: Shown on the User | Telephony | Call Setting tab. Indicates follow the settings on
the System | Telephony | Tones & Music tab.
Configuring Ring Tone Override for Groups and Incoming Call Routes
You can configure ring tone override for groups and incoming call routes. Ring Tone Override is
supported on 1400 and 9500 series phones.
Note that you can use short codes to configure a ring tone plan by using the “r” character as part of
the short code telephone number field. See Short Code Characters on page 935.
1. In Manager, select System | Telephony | Ring Tones.
2. In the Ring Tone Plan table, enter a Name for the ring tone. The Number field is populated
automatically.
3. Under Ring Tone, select one of the eight ring tones from the drop down list.
4. Once configured in this table, ring tone names can be selected from the Ring Tone
Override field at:
• Group | Group
• Incoming Call Route | Standard

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Chapter 60: Music On Hold

Each system can provide music on hold (MOH) from either internally stored files or from externally
connected audio inputs. Each system has one system source and then a number of alternate
sources (up to 3 alternate sources on IP500 V2 and 31 alternate sources on Server Edition).
You must ensure that any MOH source you use complies with copyright, performing rights and other
local and national legal requirements.
WAV Files
The system can use internal files that it stores in its non permanent memory. The WAV file
properties must be in the format listed below. If the file downloaded is in the incorrect format, it
will be discarded from memory after the download.
• PCM, 8kHz 16-bit Mono.
• Maximum length: 90 seconds on IP500 V2 systems, 600 seconds on Linux-based systems.
The first WAV file, for the system source, must be called HoldMusic.wav. Alternate source WAV
file names:
• Up to 27 IA5 characters with no spaces.
• Any file extension.
• On Linux-base systems, the filename is case sensitive.
The files, when specified by the system source or an alternate source setting, are loaded as follows:
• Following a reboot, the system will try using TFTP to download the file or files.
• The initial source for TFTP download is the system's configured TFTP Server IP Address
(System | System | LAN Settings). The default for this is a broadcast to the local subnet for
any TFTP server.
• Manager can act as a TFTP server while it is running. If Manager is used as the TFTP server,
then the wav file or files should be placed in the Manager applications working directory.

Note:
The following Manager settings are disabled by default:
- Security Settings | Unsecured Interfaces | Applications Controls | TFTP Directory
Read
- File | Preferences | Preferences | Enable BootP and TFTP Servers
• On Linux based systems, if no successful TFTP download occurs, the system automatically
looks for the files in the opt/ipoffice/tones/mohwavdir folder (disk/tones/
mohwavdir when access using file manager).

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Music On Hold

• The name of the system music .wav file should be HoldMusic.wav. The name of alternate
source .wav files should be as specified in the Alternate Sources table (System | Telephony |
Tones and Music) minus the WAV: prefix.
WAV File Download and Storage:
• If no successful TFTP download occurs:
- On IP500 V2 systems, the system automatically looks for the file in the system/primary
folder on the System SD card and downloads it from if found.
- On Linux based systems, the system automatically looks for the file in the folder opt/
ipoffice/system/primary folder (disk/system/primary when accessed using file
manager) and downloads it from there if found.
• If a music on hold file is downloaded, the system automatically write a copy of that file to its
memory card, overwriting any existing file of the same name already stored on the card.
• For files downloaded from a System SD card, the system will download the file again if the SD
card is shutdown and restarted or if files are uploaded to the card using the Embedded File
Manager.
• The system will download the file again if new files are copied to the disk or uploaded using
File Manager.
Tone
If no internal music on hold file is available and External is not selected as the System Source,
then the system provides a default tone for music on hold. The tone used is double beep
tone (425Hz repeated (0.2/0.2/0.2/3.4) seconds on/off cadence). Tone can be selected as the
System Source, overriding both the use of the external source port and the downloading of
HoldMusic.wav.
Controlling the Music on Hold Source Used for Calls
Unless specified, the System Source is used for any calls put on hold by system users. For any call,
the last source specified for the call is the one used. The following options allow the source to be
changed.
• Hunt Group Each hunt group can specify a Hold Music Source (Group | Group). That
source is then used for calls presented to the hunt group.
In a multi system network, a hunt group member will hear the music on hold (MOH) from their
local system. For example, a call comes in to site A and rings a hunt group with members from
system A and system B. If a hunt group member from system B answers a call and puts the
call on hold, the caller hears the MOH from system B.
• Incoming Call Route Each incoming call route can specify a Hold Music Source (Incoming
Call Route | Standard). That source is then used for incoming calls routed by that incoming
call route.
• Short Code The h character can be used in the Telephone Number field of short codes to
specify the hold music to associate with calls routed by that short code. The format h( X ) is
used where X is the source number. This method can be used to specify a hold music source
for outgoing calls.
Checking Music on Hold
The system short code feature Hold Music can be used to listen to the hold music sources. Dial
*34N#, replacing N with the source number 1 (System Source) or 2 to 32 (Alternate Sources).

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 System Source

Related links
System Source on page 659
Alternate Source on page 659

System Source
The first source is called the System Source. This source is numbered source 1. The possible
options for this source are:
Setting Description
WAV Use the HoldMusic.wav file. The IP Office loads the file using TFTP, or you
can directly add the file using the embedded file manager.
WAV (restart) Identical to WAV except that for each new listener, the file plays from the
beginning.
• Not supported on IP500 V2 systems.
• Cannot be used as a centralized source.
External Applicable to IP500 V2 systems. Use the audio source connected to the Audio
port on the control unit.
Tone Use a double beep tone: 425Hz, 02./0.2/0.2/3.4 seconds on/off.
• This tone is also used if the system source is set to WAV File but the
HoldMusic.wav file has not been successfully loaded.

Related links
Music On Hold on page 657

Alternate Source
You can specify alternate MOH sources on the System | Telephony | Tones and Music page.
You can assigned the alternate sources as the Hold Music Source for an Incoming Call Route
or a Group.
• That assigned MOH source overrides any current MOH source associated with the call.
• The assigned MOH source remains associated with the call as it moves around the IP Office
system. This is done using the number of the MOH source (with 1 being the number of the
default system source).
• If the call moves to another IP Office system in a multi-site network, the source with the same
number of the other system is used if also configured on that system.
•

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Music On Hold

•
IP500 V2 Alternate Sources
For IP500 V2 systems, you can specify up to 3 alternate sources. Those different types of
alternate source supported are:
Alternate Option Description
WAV:<filename> Play a specified file from its start or, if already in use, for where it is already playing.
• The <filename> parameter specifies the file to play:
- Up to 27 IA5 characters with no spaces.
- Any file extension.
- On Linux-base systems, the filename is case sensitive.
• The file location is /system/primary.
• When the source is activated, the playback resumes from where it left off last time,
instead of starting every time from the beginning.
• At any moment, all users listening to this source hear the same thing.
XTN:<extension> Play the source connected to a analog extension port on a IP500 V2 systems.
• You can set any analog extension with its Equipment Classification set as MOH
Source as an alternate source.
• The <extension> parameter specifies the analog extension's Base Extension
number. For example: XTN:224

Linux-based IP Office system

For a Linux-based IP Office system, you can specify up to 31 alternate sources. The different
types of alternate source are:
Alternate Option Description
LINE:<X>,<Y> Use an alternate source from another IP Office system in the network.
• You specify the line source using two parameters:
- <X> = The line number of the connection to the other Linux-based IP Office system
(not the outgoing group ID).
- <Y> = The MOH source number on the other Linux-based server.
• WAVRST and WAVDIRRST alternate sources are not supported.
• When the IP Office requires the source, it creates a VoIP call to the source IP Office
system. This uses call capacity from the trunk and can be subject to CAC limits.
• The IP Office drops calls to the source after 30 seconds of no use. You can change the
time using the NoUser source number HOLD_MUSIC_TIMEOUT=x. x is the number of
seconds. The range is 1 to 600 seconds.
Table continues…

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 Alternate Source

Alternate Option Description

USB:<n> Play the music streamed by a USB sound device connected to the IP Office system.
• The IP Office supported up to four USB sources. Not supported on virtual IP Office
systems.
• <n> is the logical USB device number.
- USB:1 is the first source found. This IP Office automatically uses this as the System
Source when that is set to External.
- Linux servers number additional devices sequentially. For example; USB:1, USB:2,
and so on.
• The IP Office auto-configures itself as follows:
- It selects Line input and sets the volume close to maximum.
- If it cannot identify a line input, it uses the microphone input.
• External USB sound devices are hot-pluggable. However, you must take care when
adding or removing USB sound cards, as this can change the logical number of the
device.
- When an USB MOH source is unavailable, the default MOH tone is played instead.
WAV:<filename> Play a specified file from its start or, if already in use, for where it is already playing.
• The <filename> parameter specifies the name of the file to played:
- Up to 27 IA5 characters with no spaces.
- Any file extension.
- On Linux-base systems, the filename is case sensitive.
• The file location is opt/ipoffice/system/primary.
• When the source is used, playback resumes from where it left off last time, instead of
starting from the beginning.
• At any moment, all users listening to this source hear the same thing.
WAVRST:<file> Play a specified file, beginning from its start for each caller.
• Operates similarly to WAV:<filename> above, but for each caller, playback starts
from the beginning.
WAVDIR: Play the files located in the IP Office system's mohwavdir directory.
• The directory used /disk/tones/mohwavdir (file manager access).
• Supports up to 255 files. Each file up to 10 minutes.
• The files are played in filename order (numerical, lower case, then upper case).
• At any moment, all users listening to the source hear the same thing.
• Only one WAVDIR: or WAVDIRRST: entry is supported on an IP Office system.
Table continues…

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Music On Hold

Alternate Option Description

WAVDIRRST: Play the files located in the IP Office system's mohwavdir directory, restarting from the
first file for each caller.
• Operates similarly to WAVDIR: above, but for each caller playback starts from the
beginning of the first file in the folder.
• Only one WAVDIR: or WAVDIRRST: entry is supported on an IP Office system.

Related links
Music On Hold on page 657

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Chapter 61: System Date and Time

IP Office servers can obtain their date and time either automatically from a time server or have it set
manually.
How Does the System Use the Date and Time
For files stored on memory cards the system uses the UTC time. For other activities such as call
logs, SMDR records, time display on phones; the local system time (UTC + any offsets) is used.
Related links
System Date and Time Options on page 663
Applying Daylight Saving on page 664
Checking Automatic Time and Date Operation on page 665
Manually Changing the System Date and Time on page 666

System Date and Time Options

IP Office servers can obtain their date and time either automatically from a time server or have it
set manually.
Important:
• It is strongly recommended to always use the address of an internet time server to
automatically obtain the date and time. An accurate time and date is essential to all
features that use security certificates. Manually setting the time and date should be
avoided.
Linux-based IP Office Systems
The date and time source settings are set through the server’s Platform View menus using the
Settings | System | Data and Time settings.
The supported options are:

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System Date and Time

Option Description
SNTP Use the date and time provided by an SNTP time server. The UTC time provided by the
time server is then adjusted using the server’s timezone setting.
If you have a network of servers, it is typical to set the primary server to use an external
SNTP source and all other servers are set to use SNTP from the primary server’s own
address.
Manual Enter the date and time through the platform view menu.

IP500 V2 Systems
The time and date settings for these systems are configured through their Time Setting Config
Source settings (System | System).
The supported options are:
Option Description
SNTP Get the date and time from an SNTP time server in the same way as Linux based
systems above do.
Voicemail Pro/ Get the date and time from the Windows PC running either the Voicemail Pro or IP Office
Manager Manager applications. This option requires the application to be running when the IP
Office is started and for regular time updates.
None Get the date and time from values entered via a system phone. See Manually Changing
the System Date and Time on page 666.

Related links
System Date and Time on page 663

Applying Daylight Saving

You can have the IP Office apply daylight saving time (DST) changes at certain times of the year.
How you do this depends on the type of IP Office server and the type of time source you have
configured it to use:
Server Type Description
Linux-based Daylight saving adjustments are applied to the SNTP time by defining a Location for the
Server system. The location settings include the time zone the system it is in and whether to
apply daylight saving changes for that location.
Table continues…

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 Checking Automatic Time and Date Operation

Server Type Description

IP500 V2 Server The method of applying daylight saving depends on the time source used by the server:
• SNTP/None:
The System | System menu includes settings for specifying when to apply daylight
saving.
- The system can still also use a Location to override the timezone and daylight
saving settings of the system.
• Voicemail Pro/Manager: (Obsolete)
If the system is obtaining its time from a Windows PC running Voicemail Pro or IP
Office Manager. The PC needs to be configured to apply DST to the time it provides.

Using Locations to Apply DST

In a network of IP Office systems, it may be necessary for some systems or extensions, to have
different time and date settings to match where they are physically located. This can be done by
adding Location entries to the configuration.
Each location can include a time offset from the UTC time and a set of daylight saving settings for
the location. You can then:
• Associate IP Office systems with their locations.
• Associated extension and lines with different locations if they require different settings from
their host IP Office system.
Editing the DST Calendar
Based on the system's selected timezone, the IP Office automatically adds a set of dates for when
daylight saving is applied and removed. The dates are editable.
• The current dates for applying and removing the DST settings are shown below the
Automatic DST option on the System > System and Location > Location menus.
• Each entry specifies when the IP Office should apply an additional time offset, and when the
IP Office should remove the additional time offset.
• You can use the adjacent Edit and Delete buttons to adjust the calendar entries.
• Note that the list can only include 10 entries (20 for IP Office R11.1.3.2 and higher).
- To add a new entry, you may need to delete an existing entry. After doing that, Add New
Entry appears at the bottom of the list.
Related links
System Date and Time on page 663

Checking Automatic Time and Date Operation

Operation of an IP Office server's time and date operation can be checked using the System
Status Application. Within System Status Application, the Resources > Time menu displays the
current date and time, the time source, results of the last time request and other settings.

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System Date and Time

Investigating Potential Time and Date Issues

When using an internet based time server, check the following:
1. Check the configured time server address.
2. Check routing from the server to the internet via the customer's network.
• For Linux-based servers, check the default gateway address for the customer's network
is shown in the server's Platform View menus.
• For all servers, check that the configuration includes a default IP Route to the customer
network's default gateway address.
Related links
System Date and Time on page 663

Manually Changing the System Date and Time

It is strongly recommended that IP Office systems obtain their time and date automatically from an
internet based time server. However, if configured otherwise, the following methods can be used
to change the current system time and date.
Linux-based IP Office Systems
For a Linux-based IP Office system, the system date and time can be set through the server’s
Platform View menus using the Settings | System | Data and Time settings.
IP500 V2 Systems
For IP500 V2 systems set to no time server source, date and time changes can be done though
the phone menus of a user who has been give System Phone Rights (see System Phone
Features on page 728). The user's Login Code is used to restrict access to the time and date
settings menu on the phone.
How the user accesses the date/time settings depends on the type of phone:
Phone type Details
1400, 1600, 9500, For a user with System Phone Rights, on these phones the user can set the system
9600 and J100 time and date by selecting Features | Phone User | System Administration.
Series
• This does not include the 1403, 1603 and J129 phones.
• If the system has been configured with a time server, this option can still used to
display time and date information but cannot be used to change it.

Related links
System Date and Time on page 663

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Chapter 62: Configuring Time Profiles

Time profiles are configured on Time Profile | Time Profile

Time Profiles are used by different services to change their operation when required. In most areas
where time profiles can be used, not setting a time profile is taken as meaning 24-hour operation.
Time profiles consist of recurring weekly patterns of days and times when the time profile is in effect.
Time profiles can include time periods on specified calendar days when the time profile is in effect.
Calendar records can be entered for the current and following calendar year.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Time profiles are used by the following record types.
Hunt Group:
A time profile can be used to determine when a hunt group is put into night service mode. Calls then
go to an alternate Night Service Fallback group if set, otherwise to voicemail if available or busy
tone if not.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
For automatic voice recording, a time profile can be used to set when voice recording is used.
User:
• Users being used for Dial In data services such as RAS can have an associated time profile
that defines when they can be used for that service.
• Users can be associated with a working hours and an out of hours user rights. A time profile
can then be used to determine which user rights is used at any moment.
• For automatic voice recording, a time profile can be used to set when that voice recording is
used.
• For mobile twinning, a time profile can be used to define when twinning should be used.
Incoming Call Route:
Incoming call routes can also use time profiles to specify when calls should be recorded. Multiple
time profiles can be associate with an incoming call route, each profile specifying a destination and
fall back destination.
ARS:
ARS forms use time profile to determine when the ARS form should be used or calls rerouted to an
out of hours route.

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Configuring Time Profiles

Account Code:
Account Codes can use automatic voice recording triggered by calls with particular account codes.
A time profile can be used to set when this function is used.
Auto Attendant :
Embedded voicemail auto attendants can use time profiles to control the different greetings played
to callers.
Service:
• A Service can use time profiles in the following ways:
• A time profile can be used to set when a data service is available. Outside its time profile, the
service is either not available or uses an alternate fallback service if set.
• For services using auto connect, a time profile can be used to set when that function is used.
See Service | Autoconnect.
Related links
Overriding a Time Profile on page 668

Overriding a Time Profile

You can use the Time Profile | Manual Override setting to manually override a time profile. The
override settings allow you to mix timed and manual settings.
The override options are as follows:
• Set Time Profile Active Until Next Timed Inactive
Use for time profiles with multiple intervals. Make the time profile active until the next inactive
interval.
• Set Time Profile Inactive Until Next Timed Active
Use for time profiles with multiple intervals. Make the time profile inactive until the next active
interval.
• Set Time Profile Latch Active
Set the time profile to active. Timed inactive periods are overridden and remain active.
• Set Time Profile Latch Inactive
Set the time profile to inactive. Timed active periods are overridden and remain active.
The illustration below provides an example of each override setting.

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 Overriding a Time Profile

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

A time profile can be overridden using the following methods.

• Using the Override settings on the Time Profile configuration page.
• Configure short codes for the time profile. See the description for the “Set Time Profile” short
code.
• Configure the Time Profile button action for the time profile. See the description for the “Time
Profile” button action.
Related links
Configuring Time Profiles on page 667

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Chapter 63: Applying Licenses

For a description of IP Office licenses and for information on licensing requirements, refer to the
Avaya IP Office™ Platform Solution Description document.
Related links
PLDS licensing on page 670
Web License Manager (WebLM) on page 671
Server Edition Centralized Licensing on page 672
Distributing Server Edition Licenses on page 672
Procedures for Applying Licensing on page 677
Converting from Nodal to Centralized Licensing on page 682
Migrating Licenses to PLDS on page 684

PLDS licensing
IP Office uses the Avaya Product Licensing and Delivery System (PLDS) to manage licenses.
PLDS is an online, web-based tool for managing license entitlements and electronic delivery of
software and related license files. PLDS provides customers, Avaya Partners, distributors, and
Avaya Associates with easy-to-use tools for managing license entitlements and electronic delivery
of software and related license files. Using PLDS, you can perform operations such as license
activations, license upgrades, license moves, and software downloads. You can access PLDS
from http://plds.avaya.com/.
PLDS license files
Licenses are delivered from PLDS with license files. A PLDS license file is generated for installing
on a specific machine. There are two deployment options:
• PLDS Nodal license files are generated for and installed on particular IP Office nodes.
• PLDS WebLM license files are generated for and installed on a WebLM server that can
license multiple IP Office nodes.
WebLM centralized licensing is supported in IP Office Server Edition and in IP Office Branch
deployments, but not in non-Branch deployments of IP Office Standard mode.
PLDS host ID
PLDS Nodal license files are machine specific and you must specify the host ID in the PLDS host
ID field on License | License.

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 Web License Manager (WebLM)

System Type Description

IP500 V2 Systems You can find the PLDS host ID in the Licensing tab of IP Office Manager and Web
Manager. The PLDS host ID is made of the two digits “11”, followed by the 10 digit
feature key serial number printed on the IP Office SD card. If the SD card is changed, the
PLDS host ID will also change.
IP Office Linux The PLDS host ID can be found on the server labeling, the server packaging label, and
Server the system ignition Login screen. The PLDS host ID is derived from the system ID. If the
system ID changes, the PLDS host ID will also change.
WebLM The WebLM host ID is the Mac address of the WebLM server. The WebLM host ID must
be used when generating a PLDS license file for the WebLM server in order to implement
a centralized licensing scheme for multiple IP Office systems.
The WebLM host ID can be found on the server labeling, the server packaging label, the
system ignition Login screen, and through the WebLM management interface.
In a virtual environment, the WebLM host ID is a virtual Mac address that starts with the
letter “V”.

Related links
Applying Licenses on page 670

Web License Manager (WebLM)

The Web License Manager (WebLM) is a web-based application for managing licenses. If you use
the WebLM server running on the IP Office server, then you can use IP Office Web Manager to log
in to the WebLM server by selecting Applications > Web License Manager. WebLM credentials
are managed separately from IP Office system passwords and are not part of single sign on
(SSO).
Note:
• WebLM license management is supported for Server Edition deployments and for
Enterprise Branch deployments using the System Manager WebLM server. It is not
supported for Standard Mode systems.
• When upgrading from a previous release, all systems must be running the same software
level. IP Office Server Edition does not support mixed versioning.
For more information on WebLM, see Administering standalone Avaya WebLM.
To establish communication between IP Office and the WebLM server, you must configure the
remote server profile on License | Remote Server .
Note:
When upgrading from release 9.1, the WebLM server is not started automatically. Perform the
following steps to start the WebLM server.
1. Log in to Web Manager.
2. Select Server Menu > Platform View > System.

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Applying Licenses

3. Under Services, select the WebLM server and click Start.

Related links
Applying Licenses on page 670

Server Edition Centralized Licensing

Before release 10, Server Edition deployments used nodal licensing. This type of licensing can
still be used in release 10 and higher. However, it is expected that most deployments will prefer to
centralize license management using the Avaya Web License Management (WebLM) server. The
WebLM server is automatically installed on the Server Edition Primary server. For newly installed
systems, centralized licensing is the default configuration.
All systems in the Server Edition solution must use the same License Source.
Nodal licensing
With nodal licensing, license files must be installed on each node in the system. For some
licensed features, the required license can be installed on the Server Edition Primary server and
used by all nodes in the system. However, for other licensed features, the required license must
be installed on the node where the feature is used.
Centralized licensing
As of release 10, you can use the WebLM server running on the Server Edition Primary server
to fully centralize license management. With centralized license management, all licenses are
contained in a single PLDS file uploaded to WebLM. All nodes in the solution obtain their licenses
from WebLM.
The IP Office Secondary server and Expansion systems can be configured to request licenses
directly from the WebLM server, or to use a proxy option. When configured to use the proxy
option, the license requests are sent through the IP Office Primary server, which proxies the
requests to the WebLM server. The Primary server does not allocate licenses, but only acts as a
proxy.
Systems using nodal licensing can be converted to use centralized licensing. Since PLDS license
files are generated using the host ID of the server where they reside, you must regenerate the
license file using the host ID of the WebLM server that will host the license file.
Related links
Applying Licenses on page 670

Distributing Server Edition Licenses

Note:
For a description of IP Office licenses and for information on licensing requirements, see
Avaya IP Office Platform™ Solution Description.

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 Distributing Server Edition Licenses

The License | Remote Server page displays the Reserved Licenses allocated to a Server
Edition server.
Note:
The SIP Trunk Sessions field has replaced the System | Telephony | Telephony | Max SIP
sessions setting.

PLDS File Location

How licenses are allocated depends on the location of the PLDS file. For standalone systems,
SCN deployments, and Server Edition nodal licensing, each node in the system must have a
PLDS file installed.

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Applying Licenses

PLDS File

Distributed
Primary Server Licenses

Secondary Server Expansion Expansion

Figure 1: PLDS file location for Server Edition Nodal Licensing

For Server Edition centralized licensing, the PLDS file is located on the WebLM server. The
WebLM server can be located on the Primary Server or on a remote server.

PLDS File

Primary Server WebLM

Server

Secondary Server Expansion Expansion

Figure 2: PLDS file location for Server Edition Centralized Licensing

Related links
Applying Licenses on page 670

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 Distributing Server Edition Licenses

Nodal license distribution on page 675

Centralized license distribution on page 676

Nodal license distribution

When the License Source is Local, the Reserved Licenses read-only fields indicate licenses
that are required for the currently configured features.
Nodal licensing for a Server Edition solution is based on a combination of licensing done through
the Server Edition Primary server plus some server-specific licenses. All the user specific and
system specific licenses can be managed from the Server Edition Primary server that also acts as
a licensing server. Licenses are entered into the configuration of the Server Edition Primary server
and are based on the system ID of that server.
Where a license is used to enable features, such as SIP Trunk channels, on other systems, the
Server Edition Primary server only allocates those licenses to other systems after it has met its
own license needs.
When another system loses connection to the Server Edition Primary server, any license
requirements based on those licenses entered in the Server Edition Primary server's configuration
are supported for a grace period of 30 days.
Other server specific licenses are entered into the configuration of the server requiring the feature
and are based on the System ID of that system.
License Primary server Server-specific
Server Edition
Avaya IP endpoints
Third-party IP endpoints
SIP trunk channels
IP500 universal PRI channels
Additional voicemail ports [3]
UMS Web Services [1]
Office Worker
Power User
Office Worker to Power User
upgrade
Receptionist
CTI Link Pro
Messaging TTS Pro [3]
Voicemail Pro Recording
Administrator [2] [3]
WAV User
IPSec tunneling

1. UMS Web Service licenses are for Hunt Groups only.

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Applying Licenses

2. The Voicemail Pro Recording Administrator license refers to Contact Store. Only one
license is required for a Server Edition network.
3. For deployments with dual Voicemail Pro servers, Messaging TTS Pro, Voicemail
Pro Recording Administrator, and Additional voicemail ports licenses must be on the
Secondary Server.
Related links
Distributing Server Edition Licenses on page 672

Centralized license distribution

When the license source is WebLM, the Reserved Licenses read-only fields indicate licenses
that are required for the currently configured features. Editable fields can be used to:
• Request additional licenses from the WebLM server.
• Remove licenses from the IP Office node to apply them elsewhere.
Important:
When reallocating licenses, always reduce the number on the IP Office node where they are
currently applied before applying them on another node. If you exceed the number of licenses
available, you will receive an error message.
Distribution after conversion from Nodal to Centralized licensing
• If the IP Office node needs any of the following licenses, then you must manually configure
the respective Reserved Licenses editable fields. This will allow the IP Office node to
request the licenses from the WebLM server.
- VMPro Recordings Administrators
- VMPro TTS Professional
- CTI Link Pro
Extension Reserved license setting: When the license source is Local, the setting
Extension > VoIP > Reserve License is set to None. Switching the license source to WebLM
changes the setting to Reserve Avaya IP endpoint license. If required, you must manually
change this setting to Reserve 3rd party endpoint license or Both.
License allocation in WebLM
You can use WebLM to view the licenses used by each node in IP Office Server Edition. In
the WebLM navigation pane on the left, click Licensed Products. The Acquired licenses table
displays information about the licenses acquired for each client ID. In IP Office, the WebLM client
ID for each node is displayed on the license Remote Server page.
Related links
Distributing Server Edition Licenses on page 672

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 Procedures for Applying Licensing

Procedures for Applying Licensing

Applying Licenses

Deployment
Type?

Standalone Enterprise
and SCN Branch

Uploading a PLDS
Configuring the License Server
License
for Enterprise Branch
File to IP Office

Server Edition Server Edition

Nodal Centralized

Configuring a Server Obtaining the Host ID

Edition License Source of the WebLM Server

Uploading a PLDS Installing a License File

License on the WebLM Server
File to IP Office

Configuring a Server
Edition License Source
Configuring Server
Edition Nodal Licensing

Configuring Server
Edition Centralized Licensing

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Applying Licenses

Related links
Applying Licenses on page 670
Obtaining the Host ID of the WebLM Server on page 678
Installing a License File on the WebLM Server on page 678
Configuring the Server Edition License Source on page 679
Uploading a PLDS License File to IP Office on page 679
Configuring Server Edition Nodal Licensing on page 680
Configuring Server Edition Centralized Licensing on page 680
Configuring the License Server in an Enterprise Branch Deployment on page 682

Obtaining the Host ID of the WebLM Server

The WebLM Host ID is required to generate a PLDS license file for centralized licensing. The
license file is uploaded to the WebLM server.
Procedure
1. In Web Manager, select Applications > Web License Manager.
2. Log in to WebLM.
3. In the navigation pane on the left, click Server Properties.
The Server Properties page displays the Host ID. The host ID is the MAC address of the
Server Edition Primary server.
Record the host ID.
Related links
Procedures for Applying Licensing on page 677

Installing a License File on the WebLM Server

Use Web Manager to log in to the WebLM license server and install a license file.
Before you begin
Obtain the license file from the Avaya Product Licensing and Delivery System (PLDS) Web site at
https://plds.avaya.com.
You must know the user ID and password for the WebLM server. WebLM credentials are managed
separately from IP Office system passwords and are not part of single sign on.
Procedure
1. Log in to Web Manager.
2. Select Applications > Web License Manager.
3. Log in to the WebLM server.
4. In the left navigation pane, click Install license.
5. On the Install license page, click Browse to select the license file.

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 Procedures for Applying Licensing

6. Click Install to install the license file.

WebLM displays a message upon successful installation of the license file.
If the installation is not successful, for troubleshooting information see Administering Avaya
WebLM, available on the Avaya support site at https://downloads.avaya.com/css/P8/
documents/100157154.
Related links
Procedures for Applying Licensing on page 677

Configuring the Server Edition License Source

For Server Edition deployments, the license source can be centralized or nodal.
• With centralized licensing, the PLDS license file resides on the WebLM server. The WebLM
server is the license source and all nodes in the solution receive licenses from the WebLM
server. The WebLM server can run on a remote machine or on the Primary Server.
• With nodal licensing, a PLDS license file is uploaded to each node.
All systems in the Server Edition solution must use the same license source. The license source
is defined by the configuration setting License | License | License Source. Use this procedure to
set all nodes to use the same license source.
Procedure
1. Log in to Manager.
2. On the Solution view page, on the right hand side, select Set All Nodes License Source.
3. In the Select License Source window, select either
• Local/Primary Server for nodal licensing.
• WebLM for centralized licensing.
All nodes in the solution are set to the same license source.
Related links
Procedures for Applying Licensing on page 677

Uploading a PLDS License File to IP Office

Use this procedure to upload a PLDS license file for nodal license management. Nodal license
management is used for standalone IP500 V2 systems and is an option for Server Edition
systems.
Before you begin
The PLDS license file must be on the local machine where IP Office Manager is running
Procedure
1. In IP Office Manager, select the Primary Server and then select License | License.

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Applying Licenses

2. Click Add.
3. In the Upload Files window, navigate to the license file.
4. Select the file and click Open.
Related links
Procedures for Applying Licensing on page 677

Configuring Server Edition Nodal Licensing

With nodal licensing, licenses are managed using license files installed on each node in the
system. For information on license distribution, see Distributing Nodal Licenses on page 675.
Procedure
1. In IP Office Manager, select the Primary Server and then select License | Remote Server.
2. In the Licence Source field, select Primary.

Note:
All systems in the Server Edition solution must use the same License Source. In
Manager, on the Solution page, you can select Set All Nodes License Source to
configure the setting for all nodes in the solution.
3. Enter the Server Edition Primary server IP address in the License Server IP Address
field.
4. Under Reserved Licenses, the right hand column indicates which licenses have been
reserved for this system. Use the left hand column to request additional licenses for this
system.
5. Click OK.
Licenses are displayed in the table.
6. Repeat steps 1 to 5 for the Server Edition Secondary server and all Server Edition
Expansion Systems.
Related links
Procedures for Applying Licensing on page 677

Configuring Server Edition Centralized Licensing

With centralized licensing, licenses are managed from a central WebLM server.
Before you begin
You must have a PLDS license file activated with the host ID of the WebLM server
Procedure
1. In IP Office Manager, select the Primary Server and then select License | Remote Server.
2. Ensure Licence Source is set to WebLM.

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 Procedures for Applying Licensing

Note:
All systems in the Server Edition solution must use the same License Source. In
Manager, on the Solution page, you can select Set All Nodes License Source to
configure the setting for all nodes in the solution.
3. The WebLM server can be located on the Server Edition Primary server or on a separate
server. Enter the domain name or IP address of the WebLM server in the Domain Name
(URL) field.
Note that the domain name URL must use https://.
4. If required, change the path to the WebLM server in the Path field.
5. Under Reserved Licenses, the right hand column indicates which licenses will be
automatically requested from the WebLM server. Use the left hand column to request
additional license types for this system.
6. Navigate to the Remote Server page for the Server Edition Secondary server.
7. Ensure the Licence Source is set to WebLM.
8. You can choose to enable the Enable proxy via Primary IP Office line check box.

Choice Option Choice Description

Enabled The WebLM request is sent to the WebLM server via the IP Office line
configured to the Server Edition Primary server. The line must be up and in
service
Disabled The WebLM request is sent directly to the WebLM server.

9. If Enable proxy via Primary IP Office line is enabled, enter the Server Edition Primary
server IP address in the Primary IP Address field.
10. If Enable proxy via Primary IP Office line is disabled:
a. Enter the domain name or IP address of the WebLM server in the Domain Name
(URL) field.
b. If required, change the path to the WebLM server in the Path field.
c. If required change the default Port Number.
For information on port usage see the IP Office Port Matrix document on
the Avaya support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003.
11. Click OK.
Licenses are displayed in the License | License table.
12. Repeat steps 8 to 12 for all Server Edition Expansion Systems.

Note:
In Manager, on the Solution page, you can select Set All Nodes License Source.

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Applying Licenses

Related links
Procedures for Applying Licensing on page 677

Configuring the License Server in an Enterprise Branch

Deployment
Use this procedure to configure WebLM centralized licensing where a shared PLDS license file is
installed on the WebLM server. This is the recommended method for installing license files on IP
Office systems that are centrally managed by System Manager.
For a complete description of deploying Enterprise Branch, see Deploying Avaya IP Office™
Platform as an Enterprise Branch with Avaya Aura® Session Manager.
Procedure
1. Log in to IP Office Manager and select License | Remote Server.
2. Select the Enable Remote Server check box.
The Reserved Licenses information is displayed.
3. In the Domain Name (URL), field, enter the domain name or IP address of the WebLM
server or the domain name of System Manager if the system is under System Manager
control.
4. (Optional) If a secondary System Manager is configured, enter the domain name in the
Secondary Domain Name (URL) field.
5. If required, change the path to the WebLM server in the Path field.
6. If required change the default Port Number.
For information on port usage see the IP Office Port Matrix document on
the Avaya support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003.
7. Under Reserved Licenses, the right hand column indicates which licenses will be
automatically requested from the WebLM server. Use the left hand column to request
additional licenses for this system.
Related links
Procedures for Applying Licensing on page 677

Converting from Nodal to Centralized Licensing

If you are upgrading from an earlier release, perform the procedure Migrating Licenses to
PLDS on page 684.

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 Converting from Nodal to Centralized Licensing

Note:
When upgrading from a previous release, all system must be running the same software level.
The IP Office Server Edition Solution does not support mixed versioning.
Procedure
1. You must generate a license file using the WebLM host ID. Perform the following steps to
find the WebLM host ID.
a. In Web Manager, select Applications > Web License Manager.
b. Log in to WebLM.
c. In the navigation pane on the left, click Server Properties.
The Server Properties page displays the Host ID. The host ID is the MAC address of
the Server Edition Primary server.
Record the host ID.
2. Generate a PLDS license file using the WebLM host ID.
3. Upload the license file.
a. In Web Manager, select ApplicationsWeb License Manager.
b. In the navigation pane on the left, click Install license.
c. Click Browse to select the license file.
d. Click Install to install the license file.
4. All nodes in the solution must have the same license source. To configure centralized
licensing, all nodes must have the License Source set to WebLM. You can use Manager
to set all nodes to the same license source. On the Manager Solution page, on the right
hand side, select Set All Nodes License Source and then select WebLM.
5. If you are performing this procedure after an upgrade, you must ensure that the Domain
Name (URL) field is populated on the Server Edition Primary server.
a. In Web Manager select for the Server Edition Primary server.
b. Ensure that the Domain Name (URL) field contains the domain name or IP address
of the Server Edition Primary server.
6. Reallocate the licenses as required. See Distributing Centralized Licenses on page 676.
Note that the previously install local licenses are listed as obsolete. You can use this list
to determine which licenses to request from the WebLM server. Once licenses have been
reallocated, you can delete the obsolete licenses.
Related links
Applying Licenses on page 670

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Applying Licenses

Migrating Licenses to PLDS

IP Office release 10 and higher supports only the Product Licensing and Delivery System (PLDS)
to manage license files. If you are upgrading from a previous release, you must migrate all of
your pre-R10 licenses (ADI, PLDS, mix of ADI/PLDS, virtual) to R10 PLDS licenses. The license
migration tool extracts all the licensing information from an IP Office system and saves it to a
file. This file can then be used prepare a software upgrade quote in the Avaya One Source
Configurator in order to obtain the required new PLDS R10 licenses.
For Server Edition deployments, the License Migration tool collects licensing information from
every node in the solution.
Note:
• You must use the release 10 or higher Manager client to generate the license inventory
file.
You can install Manager before upgrading to release 10. See the procedure Installing
Manager.
• License migration is supported on all IP Office modes, release 6.0 and higher.
• The license migration tool can only be used with an online configuration. The Tools >
License Migration option is disabled for offline configurations.
• The generated file can be read but must not be edited. License migration will fail if the file
has been edited.
Before you begin
Ensure all licenses are loaded on the system before performing the license migration. For Server
Edition deployments, ensure all nodes are online in order to capture the current view of systems in
the solution.
The IP Office configuration must be opened online. The License Migration tool is not available in
offline mode.
Procedure
1. Log in to Manager and select Tools > License Migration.
The Save As window opens.
2. Select a location to save the file and enter a file name.
3. Click Save.
The file is saved with a .zip extension.
Next steps
Use the file to prepare a software upgrade quote in the Avaya One Source Configurator in order to
obtain the required new PLDS R10 licenses. Once you have the PLDS license files, apply them to
the system.

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 Migrating Licenses to PLDS

Related links
Applying Licenses on page 670

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Chapter 64: Working with Templates

IP Office supports a number of template options. The settings for the following types of configuration
items can be saved as template files. New records of those types can then be created from the
template file.
• User (.usr)
• Extension (H.323, SIP, IP DECT) (.ext)
• Group (.grp)
• Service (.ser)
• Tunnel (.tnlt)
• Firewall Profile (.fpr)
• Time Profile (.tpr)
• IP Route (.ipr)
• ARS (.ars)
• Line (H.323, SIP, IP DECT) (.lne)
- The SIP trunk services from selected SIP providers are tested as part of the Avaya
DevConnect program. The results of such testing are published as Avaya Application Notes
available from the Avaya DevConnect web site (https://devconnect.avaya.com).
Related links
Saving Template files on page 686
Importing Trunk Templates on page 687
Creating a Template in Manager on page 687
Creating a New Record from a Template in Manager on page 688
Creating an Analog Trunk Template in Manager on page 689
Creating a New Analog Trunk from a Template in Manager on page 689
Applying a Template to an Analog Trunk on page 689

Saving Template files

The location used to store template files depends on the type of IP Office system.
• IP500 V2 - IP Office Manager export templates to a \manager_files\template sub-
folder of the directory where it is installed.

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 Importing Trunk Templates

• Linux-based systems: - Templates are stored on the Primary Server. When the system
configuration is opened by IP Office Manager, those templates are downloaded from server
to the \manager_files\template folder. When the configuration is saved, the templates
are uploaded back to server.
Caution:
• If you are using IP Office Manager to manage both IP500 V2 and Linux-based IP Office
systems, you need to ensure that you store the IP500 V2 templates in a directory
other than the default directory before opening any Linux-based system configuration.
When doing so, existing template in the \manager_files\template folder may be
overwritten.
Related links
Working with Templates on page 686

Importing Trunk Templates

For Standard Mode, before you can use templates from another source, they must be placed in
the Manager \Templates directory. Use this procedure to import a template from another source.
Procedure
1. Select Tools | Import Templates in Manager.
2. Browse to the current folder containing the templates that you want to import and select
that folder.
3. Click OK.
4. Any template files in the folder will be copied to the correct Manager sub-folder.
Related links
Working with Templates on page 686

Creating a Template in Manager

You can create a template from an existing record.
The options New From Template and Export as Template are available by:
• right clicking on the record type in the Navigation pane
• right clicking on a record in the Group pane
• using the Details Toolbar in the Details pane
This procedure uses the Group Pane.

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Working with Templates

Procedure
1. In the Navigation pane, select a record type.
2. In the Group pane, right click on the record on which you want to base your template and
select Export as Template.
3. The Save As window opens at the default template folder. Enter a name for the template.
A default extension is applied. For example, user templates are saved with the file
extension .usr and extension templates are saved with file extension .ext.
4. Click Save.
You can now create new records using the template.
Related links
Working with Templates on page 686

Creating a New Record from a Template in Manager

You can use a template to create additional records.
The options New from Template and Export as Template are available by:
• right clicking on the record type in the Navigation pane
• right clicking on a record in the Group pane
• using the Details Toolbar in the Details pane
This procedure uses the Group Pane.
Procedure
1. In the Navigation pane, select a record type.
2. In the Group pane, right click on the record on which you want to base your template and
select New from Template > Open from file.
Manager lists available templates stored in the default directory.
3. In the Open window, select the template file and click Open.
4. For some record types, the Import Templates window opens.
a. Enter the number of records to create in the Number of records to import field.
b. In the Start with index field, enter starting index number for the new records.
c. Click OK.
Related links
Working with Templates on page 686

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 Creating an Analog Trunk Template in Manager

Creating an Analog Trunk Template in Manager

You can create an analog trunk template from an existing trunk..
Procedure
1. In the Navigation pane, select Line.
2. In the Group pane, right click on the record on which you want to base your template and
select Generate Analog Trunk Template.
3. In the Analog Trunk Template window, you can adjust the settings if required. Click Export.
4. In the Template Type Selection window, select the Service Provider and then click Create
Template.
5. In the Browse for Folder window, select Program Files\Avaya\IP
Office\Manager\manager_files\template.
6. Click OK.
Related links
Working with Templates on page 686

Creating a New Analog Trunk from a Template in Manager

You can create a new analog trunk from a template.
Procedure
1. In the Navigation pane, right click Line and select New from Template > Open.
2. In the Open window, select a template and click Open.
3. In the Template Type Selection window, select the Service Provider and then click
Create.
Related links
Working with Templates on page 686

Applying a Template to an Analog Trunk

You can apply an analogue trunk template to existing analogue trunks.
Note:
You must reboot the system for any changes to be applied.

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Working with Templates

Procedure
1. In the group pane, right click on the analogue trunk and select Copy Setting from
Template.
2. The template and trunk selection menu is displayed.
3. In the Template Type Selection window, use the Service Provider drop-down to select the
required template.
4. Select the trunks to which you want the template to be applied.
5. Click on Copy Settings.
Related links
Working with Templates on page 686

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Chapter 65: Configuring ARS

When a dialed number matches a short code that specifies that the number should be dialled, there
are two methods by which the routing of the outgoing call can be controlled.
Routing Calls Directly to a Line
Every line and channel has an Outgoing Group ID setting. Several lines and channels can have
belong to the same Outgoing Group ID. Within short codes that should be routed via a line within
that group, the required Outgoing Group ID is specified in the short code's Line Group ID setting.
Routing Calls via ARS
The short code for a number can specify an ARS form as the destination. The final routing of the call
is then controlled by the setting available within that ARS form.
ARS Features
Feature Description
Secondary Dial The first ARS form to which a call is routed can specify whether the caller should receive
Tone secondary dial tone.
Out of Service ARS forms can be taken out of service, rerouting any calls to an alternate ARS form
Routing while out of service. This can be done through the configuration or using short codes.
Out of Hours ARS forms can reroute calls to an alternate ARS form outside the hours defined by an
Routing associated time profile.
Priority Routing Alternate routes can be made available to users with sufficient priority if the initial routes
specified in an ARS form are not available. For users with insufficient priority, a delay is
applied before the alternate routes become available.
Line Types ARS can be used with all line types.
A SIP line is treated as busy and can follow alternate routes based on the SIP line setting
Call Initiation Timeout. Previously a SIP line was only seen as busy if all the configured
channels were in use.
IP lines use the NoUser Source Number setting H.323SetupTimerNoLCR to determine
how long to wait for successful connection before treating the line as busy and following
ARS alternate routing. This is set through the IP line option Call Initiation Timeout.
Multi-Site Network Calls to multi-site extension numbers are always routed using the appropriate network
Calls trunk. ARS can be configured for multi-site network numbers but will only be used if the
network call fails due to congestion or network failure.
Main Route The ARS form 50, named "Main" cannot be deleted. For defaulted systems it is used as
a default route for outgoing calls.

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Configuring ARS

Routing Calls to ARS

1. Create the ARS form.
2. Create the required system, user or user rights short code to match the user dialing.
a. In the Telephone Number field, define the digits that will be used to match a short code
in the ARS form.
b. Use the Line Group ID field drop-down to select the ARS form required for routing the
call.
Related links
Example ARS Operation on page 692
ARS Operation on page 693
ARS Short Codes on page 695
Simple Alternate Line Example on page 696
Simple Call Barring on page 697
User Priority Escalation on page 697
Time Based Routing on page 699
Account Code Restriction on page 700
Tiered ARS Forms on page 700
Planning ARS on page 702

Example ARS Operation

The simplest example for ARS operation are the settings applied to a defaulted system. These
vary between U-Law systems and A-Law systems. For Server Edition systems refer to Server
Edition Outgoing Call Routing.
A-Law Systems
This set of defaults is applied to A-Law systems, typically supplied to locales other than North
America. The defaults allow any dialing that does not match an internal number to be routed
off-switch as follows:
1. System Short Code - ?/Dial/./50:Main
The default system short code ? will match any dialing for which no other user, user rights
or system short code match is found. This short code is set to route all the digits dialed to
ARS form 50.
2. ARS Form - 50:Main
This form contains just a single short code.
3. ?/Dial3K1/./0
This short code matches any digits passed to the ARS form. It then dials the digits out on
the first available line within line group 0 (the default outgoing line group for all lines).

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 ARS Operation

U-Law Systems
This set of defaults is applied to U-Law systems, typically supplied to locales in North America.
The defaults route any dialing prefixed with a 9 to the ARS and secondary dial tone.
1. System Short Code - 9N/Dial/N/50:Main
The default system short code 9N is used to match any dialing that is prefixed with a 9. It
passes any digits following the 9 to ARS form 50.
2. ARS Form - 50:Main
This form has secondary dial tone enabled. It contains a number of short codes which all
pass any matching calls to the first available line within line group 0 (the default outgoing
line group for all lines). Whilst all these short code route calls to the same destination,
having them as separate items allows customization if required. The short codes are:
• 11/Dial Emergency/911/0 - This short code matches an user dialing 911 for emergency
services.
• 911/Dial Emergency/911/0 - This short code matches an user dialing 9911 for
emergency services.
• 0N;/Dial3K1/0N/0 - This short code matches any international calls.
• 1N;/Dial3K1/1N/0 - This short code matches any national calls.
• XN;/Dial3K1/N/0 - This short code matches 7-digit local numbers. Note: From October
2021, telephony providers in the US have ceased routing 7-digit local numbers.
• XXXXXXXXXX/Dial3K1/N/0 - This short code matches 10-digit local numbers.
Related links
Configuring ARS on page 691

ARS Operation
The diagram below illustrates the default ARS routing applied to systems (other than Server
Edition) defaulted to the United States system locale. In summary:
• Any dialing prefixed with 9 will match the default system short code 9N.
• That short code routes calls to the default ARS form 50:Main.
• The short codes in that ARS form route all calls to an available line that has its Outgoing
Group ID set to 0.

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Configuring ARS

The table describes in more detail the process that the system has applied to the user's dialing, in
this example 91555707392200.
The user dials...
9 The Dial Delay Count is zero, so the system begins looking for short code matches in the
system and user's short codes immediately.
Since there is only one match, the 9N system short code, it is used immediately.
The 9N short code is set to route the call to the ARS form Main. It only passes those digits that
match the N part of the dialing, ie. the 9 is not passed to the ARS, only any further digits dialed
by the user.
Secondary Dial Tone is selected in the ARS form. Since no digits for ARS short code matching
have been received, secondary dial tone is played to the user.
1 Having received some digits, the secondary dial tone stops.
The ARS form short codes are assessed for matches.
The 11 and 1N; short codes are possible matches.
The 911 and 0N; short codes are not possible matches.
The XN; and XXXXXXXXXXN; short codes are also not matches because the 1N; short code is
already a more exact match.
Since there is more than one possible match, the system waits for further digits to be dialed.
Table continues…

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 ARS Short Codes

555 The 11 short code is no longer a possible match. The only match is left is the 1N; short code.
The ; in the short code tells the system to wait for the Dial Delay Time to expire after the
last digit it received before assuming that dialing has been completed. This is necessary for
line providers that expect to receive all the routing digits for a call 'en bloc'. The user can also
indicate they have completed dialing by pressing #.
707392200 When the dialing is completed, a line that has its Outgoing Group ID set to 0 (the default for
any line) is seized.
If no line is available, the alternate route settings would applied if they had been configured.

Related links
Configuring ARS on page 691

ARS Short Codes

The short codes in the default ARS form have the following roles:
Code Feature Telephone Line Group Description
Number ID
11 Dial 911 0 These two short codes are used to route
Emergency emergency calls. A Dial Emergency call is
911 Dial 911 0 never blocked. If the required line is not
Emergency available, the system will use the first available
line. Similarly, calls using Dial Emergency
ignore any outgoing call bar settings that
would be normally applied to the user.
0N; Dial 3K1 0N 0 Matches international numbers.
1N; Dial 3K1 1N 0 Matches national numbers.
XN; Dial 3K1 N 0 Matches 7 digit local numbers.
XXXXXXXXX Dial 3K1 N 0 Matches 10 digit local numbers.
XN;

ARS Short Code Settings

• Code The digits used for matching to the user dialing.
• Feature ARS short codes can use any of the Dial short code features or the Barred feature.
When a Barred short code is matched, the call will not proceed any further.
• Telephone Number The number that will be output to the line as the result of the short code
being used as the match for the user dialing. Short code characters can be used such as N to
match any digits dialed for N or X in the Code.
• Line Group ID The line group from which a line should be seized once short code matching
is completed. Another ARS form can also be specified as the destination.
• Locale Not used for outgoing external calls.

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Configuring ARS

• Forced Account Code If enabled, the user will be prompted to enter a valid account
code before the call can continue. The account code must match one set in the system
configuration.
Related links
Configuring ARS on page 691

Simple Alternate Line Example

Using the default ARS settings, despite having several short codes in the ARS form, all outgoing
calls are actually routed the same way using the same trunks. However, by having separate short
codes for different call types present, it is easy to change the routing of each call type if required.
For this example, the customer has separate sets of lines for local calls and for national/
international calls. These have been configured as follows:
• The lines for local and emergency calls have been left with the default Outgoing Group ID of
0.
• The lines for national and international calls have been set with the Outgoing Group ID of 1.
The default ARS can be configured to match this by just changing the Line Group ID settings of
the default ARS short codes to match.

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 Simple Call Barring

Related links
Configuring ARS on page 691

Simple Call Barring

All ARS short codes use one of the Dial short code features. The exception is the Barred short
code feature. This can be selected for ARS short codes that match dialing that is not allowed.
In the example below, any user dialing an international number will be routed to the Barred short
code. This prevents the dialing of external numbers prefixed with 0.

To restrict a user from making any outgoing external calls, use the user's Outgoing Call Bar option.
Related links
Configuring ARS on page 691

User Priority Escalation

User priority can be used to alter call routing when the required route is not available.
In this example, international calls are initially targeted to seize a line in outgoing line group
1. However an alternate route has been defined which will be used if no line in line group 1

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Configuring ARS

is available. The fallback ARS form allows international calls to seize a line from line group 0.
Whether this is done immediately or after a delay is set by whether the users priority is high
enough.

Related links
Configuring ARS on page 691

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 Time Based Routing

Time Based Routing

Time profiles can be used to switch call routing from one ARS form to another.
In the example below, a time profile has been define that sets the hours for normal operation.
Outside the times set in the time profile, the other ARS form is used. This other ARS form only
allows local and emergency calls.

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Configuring ARS

Related links
Configuring ARS on page 691

Account Code Restriction

The short codes within an ARS form can be individually set to require an account code before
allowing any call that matches it to proceed.
In the example below, the short code for international calls has been set to require the user to
enter an account code. A valid account code must be dialed to continue with the call.

If a user should always enter an account code to make any external call, the user option Force
Account Code should be used.
Related links
Configuring ARS on page 691

Tiered ARS Forms

It is possible for an ARS short code in one form to have another ARS form as its destination.
Dialing that matches the short code is then subject to further matching against the short codes in
the other ARS form.

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 Tiered ARS Forms

In the example below, the user wants different routing applied to international calls based on the
country code dialed. To do that in the default ARS form would introduce a large number of short
codes in the one form, making maintenance difficult.
So the short code matching calls with the international dialing prefix 0 has been set to route
matching calls to another ARS form. That form contains short codes for the different country
dialing codes of interest plus a default for any others.

Related links
Configuring ARS on page 691

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Configuring ARS

Planning ARS
Using the methods shown in the previous examples, it is possible to achieve ARS that meets most
requirements. However the key to a good ARS implementation is planning.
A number of questions need to be assessed and answered to match the system's call routing to
the customer's dialing.
What What numbers will be dialed and what needs to be output by the system. What are the
different call tariffs and the dialing codes.
Where Where should calls be routed.
Who Which users should be allowed to use the call routes determined by the previous questions.
When When should outgoing external calls be allowed. Should barring be applied at any particular
times? Does the routing of calls need to be adjusted for reasons such as time dependant call
tariffs.
Related links
Configuring ARS on page 691

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Chapter 66: Call Barring

Related links
Applying Call Barring on page 703
Overriding call barring on page 704

Applying Call Barring

Call barring can be applied in a number of ways.
Barring a User From Receiving Any External Calls:
For each user, the Incoming Call Bar setting (User | Telephony | Supervisor Settings ) can be
selected to stop that user from receiving any external calls.
Barring a User From Making Any External Calls:
For each user, the Outgoing Call Bar setting (User | Telephony | Supervisor Settings) can be
selected to stop that user from making any external calls.
Barring Particular Numbers/Number Types:
System short codes are used to match user dialing and then perform a specified action. Typically
the action would be to dial the number to an external line. However, short codes that match the
dialing of particular numbers or types of numbers can be added and set to another function such
as Busy. Those short codes can be added to a particular user, to a User Rights associated with
several users or to the system short codes used by all users.
The system allows short codes to be set at user, user rights, system and least cost route. These
have a hierarchy of operation which can be used to achieve various results. For example a system
short code for a particular number can be set to busy to bar dialing of that number. For a specific
user, a user short code match to the same number but set to Dial will allow that user to override
the system short code barring.
Using Account Codes:
The system configuration can include a list of account codes. These can be used to restrict
external dialing only to users who have entered a valid account code.
• Forcing Account Code Entry for a User: A user can be required to enter an account code
before the system will return dialing tone. The account code that they enter must match a
valid account code stored in the system configuration. The setting for this is Force Account
Code (User | Telephony | Supervisor Settings)

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Call Barring

• Forcing Account Code Entry for Particular Numbers: Each system short code has a
Force Account Code option. Again the account code entered must match a valid account
code stored in the system configuration. for the call to continue.
Barring External Transfers and Forwards:
A user cannot forward or transfer calls to a number which they cannot normally dial. In addition
there are controls which restrict the forwarding or transferring of external calls back off-switch. See
Off-Switch Transfer Restrictions on page 784.
Related links
Call Barring on page 703

Overriding call barring

When a system or user short code is configured to bar outgoing calls, you can override call
barring. Typically, this configuration is used for a phone in a shared or public area. By default,
the phone has outgoing calls barred. The administrator can override call barring for specific dialed
numbers by entering numbers with a record in the external directory. When the dialed number
exists in the external directory and the Directory Overrides Call Barring setting is enabled, call
barring is overridden.
The System Directory entries must use the format (shortcode)number. For example, if the number
to dial is 61234, where 6 is the shortcode used to dial externally and 1234 is the number, the
System Directory entry must be (6)1234. If the dial shortcode contains a name string rather than
digits, then Directory Overrides Call Barring will not work.
The Directory Overrides Barring setting is located on the System | Telephony | Telephony tab.
For information on the directory, see the description for the System | Directory Services tab.
Server Edition configuration
For Server Edition deployments, the Directory Overrides Barring must be enabled on each
node. It is not a system wide setting.
For example, if the Primary Server uses an IP500 V2 expansion system as an ISDN gateway,
Directory Overrides Barring must be enabled on the Primary Sever for Primary Server users
dialing on external ISDN lines. For the IP500 V2 expansion users, Directory Overrides Barring
must be enabled on the IP500 V2 expansion system.
It is recommend that the short code configured to dial externally on ISDN lines be the same on all
nodes. For example, if Primary Server users and IP500 V2 expansion users want to reach PSTN
number 123456789 on ISDN lines, configure the dial codes as follows.
• Primary Server: 6N/Dial/6N/XX (XX is the line group ID for the SCN line)
• IP500 V2 expansion: 6N/Dial/N/YY (YY is the line group ID for ISDN line)
• Directory Entry number defined on Primary Server: (6)123456789
Related links
Call Barring on page 703

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Chapter 66: Configuring authorization
codes

Note:
For Release 9.1 and higher, you can no longer associate Authorization Code entries with User
Rights. Authorization Code configured in that way are removed during the upgrade.
Authorization codes are enabled by default.
A user dials a number that matches a short code set to Force Authorization Code. The user is
prompted to enter an authorization code.
They dial their authorization code. If a matching entry is found in Authorization Codes records the
system checks the corresponding user. Note that the user checked does not necessarily need to be
connected with the user dialing or the user whose extension is being used to make the call.
The dial string is checked against the short codes with the matching user. If it matches a dial short
code or no short code the call is allowed, otherwise it is blocked. Note that the short code is not
processed, it is just checked for a match. If multi-tier authorization codes are required there must be
blocking (busy) short codes (or a wild card '?' )
Example:
A restaurant has a number of phones in publicly accessible areas and want to control what calls can
be made by staff. Staff must not be able to dial long distance numbers. staff should be able to dial
local and cell phone numbers.
ARS Table
In the Main (50) ARS table, add the following short codes:
• 044XXXXXXXXXX/Dial/044N/
• 01XXXXXXXXXX/Dial/01N/Force Auth Code checked
Authorization Codes
Configure an authorization code for each staff member that is allowed to make long distance calls. For
example, for staff members Alice and Bob:
AuthCode: 2008 - Alice
AuthCode: 1983 - Bob

It is recommended to use short codes that use X characters to match the full number of characters
to be dialed. That ensures that authorization code entry is not triggered until the full number has
been dialed rather than mid-dialing. For example 09 numbers are premium rate in the UK, so you

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Configuring authorization codes

would create a 09XXXXXXXXX/Dial/N short code set to Forced Authorization. In the associated
user or user right short code it is recommended to use 09N type short codes.
System short codes that route to ARS will not have their Force Authorization Code setting used.
However short codes within an ARS table will have their Force Authorization Code setting used.
Forcing Authorization Codes
There are two methods to force a user to enter an authorization code in order to complete dialing an
external call.
• To Force Authorization Codes on All External Calls A user can be required to enter an
authorization code for all external calls. This is done by selecting Force Authorization Code
(User | Telephony | Supervisor Settings).
• To Force Authorization Codes on Specific Calls To require entry of an authorization code
on a particular call or call type, the Force Authorization Code option should be selected in
the short code settings. This can be used in user or system short codes in order to apply its
effect to a user or all users respectively. You need to ensure that the user cannot dial the same
number by any other method that would by pass the short code, for example with a different
prefix.
Related links
Entering an Authorization Code on page 706

Entering an Authorization Code

Where possible, when an authorization code is required, the user can enter it through their phones
display. However, this is not possible for all type of phone, for example it is not possible with
analog phones and Avaya XX01 or XX02 phones. The users of these device must either enter
the authorization code using a short code set to the Set Authorization Code feature immediately
before making the call.
When entry of an authorization code is triggered, the user can enter any authorization code with
which they are directly associated.
Note the following.
• If authorization code entry is setup for a particular number, calls forwarded or transferred to
that number will also trigger authorization code entry.
• On systems using line appearances to BRI trunk channels to make outgoing calls,
authorization code entry may not be triggered. This can be resolved by adding a short code
such as [9]XN;/Dial/XN/0 (adjust the prefix and line group as necessary).
Related links
Configuring authorization codes on page 705

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Chapter 66: Preventing Toll Bypass

Use this procedure to prevent toll bypass in Enterprise Branch and Small Community Network
(SCN) deployments. Toll bypass is prevented by only allowing PSTN calls where the originating
location and terminating location are the same.
The location of non-IP lines is the same as the system location. If an IP address is not resolved
to any location, then that device is assumed to be in the system location. The location of public IP
lines must be configured to same as PSTN termination location.
The Location field for extensions with simultaneous login must be automatic and the location tab
must be properly configured for the IP range.
Enterprise Branch deployments: All the distributed users must be in the same location as
system location. Users registering from a location different from the system location are not
supported.
Procedure
1. In the navigation pane on the left, select System.
2. In the details pane, click the Telephony tab.
3. Under Telephony, click the Telephony tab.
4. On the Telephony tab:
a. Click the check box to turn Restrict Network Interconnect on.
b. Click the check box to turn Include location specific information on.
Setting the two configuration setting on the Telephony tab adds a Network Type field
to the configuration settings for each trunk.
5. For Enterprise Branch deployments, open the SM Line | Session Manager tab. For SCN
deployments, open the IP Office Line | Line tab.
6. If the line is a PSTN trunk (includes SIP), set Network Type to Public. If the line is an
enterprise trunk, set the Network Type to Private.
7. If the Network Type is Private, the Include location specific information field is available.
If the line is connected to an Avaya Aura® system release 7.0 or higher, or an IP Office
release 9.1 or higher, set Include location specific information to On.
Related links
Configuring unknown locations on page 708

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Preventing Toll Bypass

Configuring unknown locations

Use this procedure to configure extensions where the location is unknown.
Procedure
1. In the navigation pane, select Location.
2. Enter a Location Name.
3. Set Parent Location for CAC to Cloud.
4. In the Extension > Extn tab, set the Location field to the location defined in step 2.
Related links
Preventing Toll Bypass on page 707

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Chapter 66: Configuring Call Admission
Control

Call Admission Control (CAC) is a method of controlling system resources using defined locations.
Calls into and out of each location are allowed or disallowed based upon configured call constraints.
In Manager, use the Location tab to define a location and configure the maximum calls allowed for
the location.
Related links
Manager location tab on page 709
Assigning a network entity to a location on page 710
System actions at maximum call threshold on page 710
Example on page 711

Manager location tab

Configuring location settings
On the Manager Location tab, set the following parameters for a location:
• Location Name
• Subnet Address
• Subnet Mask
Configuring Call Admission Control settings
On the Manager Location tab, set the following CAC parameters:
• Internal Maximum Calls: Calls that pass from the location to another configured location.
• External Maximum Calls: Calls that pass from the location to an unmanaged location.
• Total Maximum Calls: The total internal and external calls permitted.
Related links
Configuring Call Admission Control on page 709

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Configuring Call Admission Control

Assigning a network entity to a location

The Location field is a drop down list of locations defined on the Location tab. Network entities
are assigned to a location using the Location field on the following Manager tabs.
• System
• Extension
• SIP Line | VoIP
• H323 Line | VoIP
The following default settings are applied.
• Each IP Office system can be configured with a defined location. For Server Edition
deployments, the configuration of locations is done solution wide. All IP Office systems in
the solution share the same location configuration.
• Digital phones default to the system location.
• The default setting for IP phones is Automatic. Phones registering from a subnet matching
that of a location will be treated as within that location. Otherwise, the phone is assigned the
same location as the system. Cloud can be used for phones whose Location is variable or
unknown.
• IP Lines default to Cloud.
Related links
Configuring Call Admission Control on page 709

System actions at maximum call threshold

• A congestion alarm is raised.
• Calls that exceed the CAC maximum values are not allowed.
• Calls from extensions to public trunks through Alternate Route Selection (ARS) are queued
and display Waiting for Line.
• Calls from extensions to public trunks which do not route through ARS receive a fast-busy
tone and display Congestion.
• Idle phones display Emergency/Local calls only.
• Alternative routing to a local PSTN gateway follows ARS priority escalation rules.
• SIP calls that would exceed call limits and have no other targets are declined with
cause=486 or cause = 503.
Allowed calls
When CAC limits have been reached, the following calls are allowed.
• Emergency calls are always allowed.

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 Example

• Established calls are never torn down to achieve limits.

• A phone on a remote site that parks a call is always allowed to retrieve it.
• Request Coaching Intrusion calls are allowed.
Related links
Configuring Call Admission Control on page 709

Example
The example configuration has four locations.
Location Max Calls
HQ 20
RS1 5
RS2 10
RS3 15
+Cloud unlimited

SIP Line 10 and SIP Line 11 are configured with 20 channels.

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Configuring Call Admission Control

Notes
• Calls between locaton RS1 and SBC113 do not increment the call count for HQ.
• The HQ call count includes calls across the HQ boundary which anchor media inside HQ.
SBC113 and SBC 114 are both included.
• The HQ maximum calls value is separate and complementary to the individual trunk call
maximum.
• Incoming calls from SIP to RS1 (direct media) only need to verify that the RS1 location
maximum call value is not exceeded.
• SIP calls that are not allowed to RS1 may go to HQ voicemail if the HQ call limit is not
exceeded.
Related links
Configuring Call Admission Control on page 709

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Chapter 67: Configure User Settings

Related links
User Management Overview on page 713
Configuring Gmail Integration on page 715
Call Intrusion on page 716
Call Tagging on page 719
Call Waiting on page 719
Call Barring on page 720
Centralized Call Log on page 721
Centralized Personal Directory on page 721
Account Code Configuration on page 722
Malicious Call Tracing (MCID) on page 724
Twinning on page 724
Private Calls on page 727
System Phone Features on page 728
The 'No User' User on page 729

User Management Overview

Users are the people who use the system. They do not necessary have to be an extension user,
for example users are used for RAS dial in data access. In addition, more users can be created
than there are extensions, with users logging in to an extension when they want to receive calls.
By default, a user is automatically created to match each extension. They are numbered from 201
upwards and the first 16 are placed in the hunt group Main (200), which is the default destination
for incoming calls.
Terminology
Standard User: A standard user.
Centralized User: Centralized users can be provisioned for enterprise branch deployments.
No User: Used to apply settings for extensions which currently have no associated user. The
SourceNumbers settings of the NoUser user is used to configure a number of special options.
These are then applied to all users on the system.
Remote Manager: Used as the default settings for dial in user connections.

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Configure User Settings

Hot Desking User: Users with a Login Code can move between extensions by logging in and off.
Deleting a User
When a user is deleted, any calls in progress continue until completed. The ownership of the call
is shown as the NoUser user. Merging the deletion of a user causes all references to that deleted
user to be removed from the system.
Changing a User's Extension
Changing a user's extension number automatically logs the user in on the matching base
extension if available and the user doesn't have Forced Login enabled. If Forced Login is
enabled, then the user remains on the current extension being used until they log out and log
in at the new extension.
Note that changing a user's extension number affects the user's ability to collect Voicemail
messages from their own extension. Each user's extension is set up as a "trusted location" under
the Source Numbers tab of the User configuration form. This "trusted location" allows the user to
dial *17 to collect Voicemail from his own extension. Therefore if the extension number is changed
so must the "trusted location".
The following related configuration items are automatically updated when a user extension is
changed:
• User, Coverage and Bridged Appearance buttons associated with the user.
• Hunt group membership (disabled membership state is maintained).
• Forwards and Follow Me's set to the user as the destination.
• Incoming call routes to this destination.
• Dial in source numbers for access to the user's own voicemail.
• Direct call pickup buttons are updated.
• The extension number of an associated extension is updated.
Server Edition User Management
In a Server Edition network, individual users are still added to the configuration of a particular
server. Typically they are added to the configuration of the server that hosts the user's physical
extension or supports their main place of work. That server is treated as the host system for the
user. However, once a user is added to the configuration of a particular system, you can use
Manager and Web Manager to manage all users in the Server Edition solution.
Centralized User Management
Centralized Users are provisioned for enterprise branch deployments. Centralized Users
are registered with Session Manager and are able to utilize telephony features provided by
Communication Manager. The Centralized User profile is applicable to both SIP and analogue
extensions. For more information, see Administering Centralized Users for an IP Office™ Platform
Enterprise Branch. The following requirements must be met when provisioning a centralized user:
• An SM line must be configured on the system.
• The user must be provisioned with an existing extension.
• The extension Base Extension value must match the centralized extension value.
• Centralized users must be configured with a password for SIP registration on Session
Manager. The password is set in User | Telephony | Supervisor Settings | Login Code field.

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 Configuring Gmail Integration

Related links
Configure User Settings on page 713

Configuring Gmail Integration

You can integrate the Google Gmail application into Voicemail Pro in order to use a Gmail account
for voicemail to email functions. The supported functions are:
• Forward: Voicemail messages are sent as email to the Gmail account of a user. Users can
use Gmail to retrieve and manage emails.

• Copy: Copies of voicemail messages are sent as email to the Gmail account of a user. The
message is also stored locally on the Voicemail Pro server.
• Alert: An email is sent to the Gmail account of a user indicating the arrival of a new
voicemail.
For the forwarding function:
• Up to 250 users are supported.
• The maximum message length is 7 minutes or 14 minutes when using companded.
• Messages can be accessed using Visual Voice but not one-X Communicator.

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Configure User Settings

User's Gmail Account

HTTP RestFul Access

Google API Framework

Gmail Service API

Primary Server

Voicemail Pro Server

Related links
Configure User Settings on page 713

Call Intrusion
The IP Office system supports several different methods for call intrusion. The method used
affects which parties can hear each other. Intrusion features are supported across a multi-site
network.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.

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 Call Intrusion

Warning:
• Listening to a call without the other parties being aware is subject to local regulations.
You must ensure that you have complied with the local regulations. Failure to do so can
result in penalties.
In the examples below, A has called or is calling IP Office user B. A can be internal or external.
User C invokes one of the call intrusion methods targeting user B.
Description Privacy Settings Used
User Target
Can Cannot Private
Intrude Be Call
Intruded
Call Listen ✓ ✓ ✓
Hear another user's call without being heard.
• Monitoring can include a tone heard by all parties. This is controlled by
the Beep on Listen setting (System > Telephony > Tones & Music).
• Call Listen can only intrude on calls to users in a user's Monitor
Group (User > Telephony > Supervisor Settings).
Call Intrude ✓ ✓ ✓
Intrude on the existing connected call of the another user. All call parties
are put into a conference and can talk to and hear each other.
• A Call Intrude attempt to a user who is idle becomes a Priority Call.

Call Steal ✓ ✓ ✓
Take a connected or alerting call from another user.
• If the target has multiple alerting calls, the function steals the longest
waiting call.
• If the target has a connected call and no altering calls, the function
steals the connected call. This is subject to the Can Intrude setting of
the Call Steal user and the Cannot Be Intruded setting of the target.
• If no target is specified, the function attempts to reclaim the user's
last ringing or transferred call if it has not been answered or gone to
voicemail.
• Stealing a video call changes the call to an audio call.
• R11.1 FP2 SP4 and higher: The shortcode for this feature can be
used with the user's own extension number. That enables twinned and
simultaneous device users to move a connected call from another one
of their devices. This usage ignores the user's privacy and intrusion
settings.
Table continues…

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Configure User Settings

Description Privacy Settings Used

User Target
Can Cannot Private
Intrude Be Call
Intruded
Dial Inclusion ✓ ✓ ✓
Temporarily interrupt another user's call to talk to them. Their current
call is held whilst you talk. When you hang-up, the original calls is
reconnected.
• You and the user can talk but cannot be heard by the other party.
• You can intrude on a user in a conference. The conference continues
without the user.
• During the intrusion, all parties hear a repeated intrusion tone.
• Attempting to hold a dial inclusion call ends the intrusion.
• You cannot park an inclusion call.
Whisper Page ✓ ✓ ✕
Intrude on another user and be heard by them without interrupting or
being able to hear their existing call.
• You can use whisper page to talk to a user who has enabled private
call.

Coaching Intrusion ✓ ✓ ✓
Intrude on another user's call and to talk to them without being heard by
the other call parties to which they can still talk.
• Example: When C intrudes on B, they can hear A and B, but only B can
be hear C.

Request Coaching Intrusion ✓ ✓ ✓

Request a coaching intrusion.
• Example: B requests coaching from C. When C responds, they can
hear A and B, but only B can hear C.

Appearance Buttons ✕ ✓ ✓
Users can press appearance buttons indicating 'in use elsewhere' to join
the call.
• The Can Intrude setting of the user is not used.
• This feature uses the Cannot Be Intruded setting of the call's longest
present internal user.

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 Call Tagging

Related links
Configure User Settings on page 713

Call Tagging
Call tagging associates a text string with a call. That string remains with the call during transfers
and forwards. That includes calls across a multi-site network.
On Avaya display phones, the text is shown whilst a call is alerting and is then replace by the
calling name and number when the call is connected. On analog phones with a caller ID display,
the tag text replace the normal caller information.
Applications such as SoftConsole display any call tag associated with a call. If the call is parked,
the tag is shown on the call park slot button used. A call tag can be added when making a call
using SoftConsole or one-X Portal. A tag can be added to a call by an Incoming Call Route or by
an Voicemail Pro Assisted Transfer action.
Related links
Configure User Settings on page 713

Call Waiting
Call waiting allows a user who is already on a call to be made aware of a second call waiting to be
answered.
User Call Waiting
Call waiting is primarily a feature for analog extension users. The user hears a call waiting tone
and depending on the phone type, information about the new caller may be displayed. The call
waiting tone varies according to locale.
For Avaya feature phones with multiple call appearance buttons, call waiting settings are ignored
as additional calls are indicated on a call appearance button if available.
To answer a call waiting, either end the current call or put the current call on hold, and then
answer the new call. Hold can then be used to move between the calls.
Call waiting for a user can be enabled through the system configuration (User | Telephony | Call
Settings | Call Waiting On) and through programmable phone buttons.
Call waiting can also be controlled using short codes. The following default short codes are
available when using Call Waiting.
*15 - Call Waiting On Enables call waiting for the user.
*16 - Call Waiting Off Disables call waiting for the user.

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Configure User Settings

*26 - Clear Call and Answer Call Waiting Clear the current call and pick up the waiting call.
Hunt Group Call Waiting
Call waiting can also be provided for hunt group calls. The hunt group Ring Mode must be
Collective Call Waiting.
On phones with call appearance buttons, the call waiting indication takes the form of an alert on
the next available call appearance button. On other phones, call waiting indication is given by a
tone in the speech path (the tone is locale specific).
The user's own Call Waiting setting is overridden when they are using a phone with call
appearances. Otherwise the user's own Call Waiting setting is used in conjunction with the hunt
group setting.
Related links
Configure User Settings on page 713

Call Barring
Call barring can be applied in a range of ways.
Barring a User From Receiving Any External Calls
For each user, User > Telephony > Supervisor Settings > Incoming Call Bar can be selected
to stop that user from receiving any external calls.
Barring a User From Making Any External Calls
For each user, User > Telephony > Supervisor Settings > Outgoing Call Bar can be selected
to stop that user from making any external calls.
Barring Particular Numbers/Number Types
The system allows short codes to be set at user, user rights, system and least cost route. These
have a hierarchy of operation which can be used to achieve various results. For example a system
short code for a particular number can be set to busy to bar dialing of that number. For a specific
user, a user short code match to the same number but set to Dial will allow that user to override
the system short code barring.
System short codes are used to match user dialing and then perform a specified action. Typically
the action would be to dial the number to an external line. However, short codes that match the
dialing of particular numbers or types of numbers can be added and set to another function such
as Busy. Those short codes can be added to a particular user, to a User Rights associated with
several users or to the system short codes used by all users.
Using Account Codes
The system configuration can include a list of account codes. These can be used to restrict
external dialing only to users who have entered a valid account code.
• Forcing Account Code Entry for a User - A user can be required to enter an account
code before the system will return dialing tone. The account code that they enter must

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 Centralized Call Log

match a valid account code stored in the system configuration. The setting for this is User >
Telephony > Supervisor Settings > Forced Account Code.
• Forcing Account Code Entry for Particular Numbers - Each system short code has a
Force Account Code option. Again the account code entered must match a valid account
code stored in the system configuration. for the call to continue.
Barring External Transfers and Forwards
A user cannot forward or transfer calls to a number which they cannot normally dial. In addition
there are controls which restrict the forwarding or transferring of external calls back off-switch. See
Off-Switch Transfer Restrictions on page 784.
Related links
Configure User Settings on page 713

Centralized Call Log

The IP Office stores a centralized call log for each user:
• The user's centralized call log is stored by the IP Office system on which the user is
configured. If the user is logged in on another system, new call log records are sent to
the user's home IP Office system, but using the time and date on the IP Office system where
the user is logged in.
• Each user's centralized call log can contain 30 (IP500 V2) or 60 (Server Edition) call records.
Each new call record replaces the oldest previous record when it reaches the limit.
• By default, the centralized call log is displayed on Avaya phones with a fixed Call Log
or History button (1400, 1600, 9500, 9600, J100 Series) and in the one-X Portal, Avaya
Workplace Client, and IP Office User Portal applications.
• The centralized call log moves with the user as they log onto different phones and IP Office
applications.
• The missed call count is updated per unique caller, not per call.
Adjusting Call Log Operation
The operation of the centralized call log is control by the System > Telephony > Call Log and
User > Telephony > Call Log settings.
Related links
Configure User Settings on page 713

Centralized Personal Directory

Each system user is able to have up to 250 personal directory records stored by the system. A
user's personal directory is also usable with 1400, 1600, 9500, 9600 and J100 Series (including

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Configure User Settings

J129) phones with a CONTACTS button. The user can view these records and use them to make
calls.
Phone users can edit their personal directory records through the phone. The user personal
directory records can be edited by administrator through the User > Personal Directory menu in
IP Office Manager and IP Office Web Manager. Users can edit their personal directory through
their phone or using the user portal application.

When the user hot desks to another phone that supports the centralized personal directory, their
personal directory records become accessible through that phone. That also includes hot desking
to another system in the network.
Related links
Configure User Settings on page 713

Account Code Configuration

Forcing Account Code Entry for Specific Numbers
You can make entry of an account code a requirement for any dialing that matches a particular
short code. This is done by ticking the Force Account Code option found in the short code
settings.
Note that the account code request happens when the short code match occurs. Potentially this
can be in the middle of dialing the external number, therefore the use of X wildcards in the short
code to ensure full number dialing is recommended.
Entering Account Codes
The method for entering account codes depends on the type of phone being used. Refer to the
relevant telephone User's Guide for details.

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 Account Code Configuration

Account Code Button:

The Account Code Entry action (User | Button Programming | Emulation | Account Code
Entry) and Set Account Code action (User | Button Programming | Advanced | Set | Set
Account Code) can be assigned to a programmable button on some phones. They both operate
the same. The button can be preset with a specific account code or left blank to request account
code entry when pressed. The button can then be used to specify an account code before a call or
during a call.
Setting an Account Code using Short Codes:
The Set Account Code feature allows short codes to be created that specify an account code
before making a call.
Show Account Code Setting :
This setting on the System | Telephony | Telephony tab controls the display and listing of system
account codes.
When on and entering account codes through a phone, the account code digits are shown while
being dialed.
When off and entering account codes through a phone, the account code digits are replaced by s
characters on the display.
Server Edition Account Code Management
Accounts codes configured on Server Edition are shared by all systems in the network.
Related links
Configure User Settings on page 713
Setting a User to Forced Account Code on page 723

Setting a User to Forced Account Code

Procedure
1. Receive the system configuration if one is not opened.
2. In the left-hand panel, click User. The list of existing user is shown in the right-hand panel.
3. Double-click the required user.
4. Select the Telephony tab.
5. Tick the Force Account Code option.
6. Click OK.
7. Merge the configuration.
Related links
Account Code Configuration on page 722

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Configure User Settings

Malicious Call Tracing (MCID)

MCID (Malicious Caller ID) is an ISDN feature. It is supported on BRI and PRI trunks to ISDN
service provider who provide MCID.
When used, it instructs the ISDN exchange to perform a call trace on the user's current call and
to keep a record of the call trace at the exchange for the legal authorities. Trace information is not
provided to or displayed by the system or system phones.
The use of MCID is subject to local and national legal requirements that will vary. The feature may
also not be enabled until specifically requested from the service provider. You should consult with
your ISDN service provider and with appropriate legal authorities before attempting to use MCID.
Note:
Currently, in Server Edition network, MCID is only supported for users using an MCID button and
registered on the same IP500 V2 Expansion system as the MCID trunks.
Activating MCID
1. Liaise with the ISDN Service Provider MCID should not be used without first confirming
its usage with the ISDN service provider.
2. Enabling MCID Call Tracing on a Line BRI and PRI lines include a Support Call Tracing
Option which by default is off.
3. Enabling MCID Call Tracing for a User Each user has a Can Trace Calls (User |
Telephony | Supervisor Settings) option. This option is off by default.
4. Providing an Active MCID Control The user needs to be provided with a mechanism to
trigger the MCID call trace at the exchange. This can be done using either a short code or
a programmable button.
• MCID Activate Button The action MCID Activate (Advanced | Miscellaneous | MCID
Activate) can be assigned to a programmable buttons. It allows a malicious call trace to
be triggered during a call.
• MCID Activate Short Codes The feature MCID Activate can be used to create a short
code to triggering a malicious call trace.
Related links
Configure User Settings on page 713

Twinning
Twinning allows a user's calls to be presented to both their current extension and to another
number. The system supports two modes of twinning:

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 Twinning

Internal Mobile
Twinning Destination Internal extensions only External numbers only.
Supported in All locales. All locales.
License Required No No

User BLF indicators and application speed dials set to the primary user will indicate busy when
they are connected to a twinned call including twinned calls answered at the mobile twinning
destination.
Do Not Disturb and Twinning
Mobile Twinning
Selecting DND disables mobile twinning.
Internal Twinning
• Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.
• Logging out or setting do not disturb at the secondary only affects the secondary.
Do Not Disturb Exceptions List
For both types of twinning, when DND is selected, calls from numbers entered in the user's Do Not
Disturb Exception List are presented to both the primary and secondary phones.
Internal Twinning
Internal twinning can be used to link two system extensions to act as a single extension. Typically
this would be used to link a users desk phone with some form of wireless extension such as a
DECT or WiFi handset.
Internal twinning is an exclusive arrangement, only one phone may be twinned with another.
When twinned, one acts as the primary phone and the other as the secondary phone. With
internal twinning in operation, calls to the user's primary phone are also presented to their twinned
secondary phone. Other users cannot dial the secondary phone directly.
• If the primary or secondary phones have call appearance buttons, they are used for call
alerting. If otherwise, call waiting tone is used, regardless of the users call waiting settings. In
either case, the Maximum Number of Calls setting applies.
• Calls to and from the secondary phone are presented with the name and number settings of
the primary.
• The twinning user can transfer calls between the primary and secondary phones.
• Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.
• Logging out or setting do not disturb at the secondary only affects the secondary.
• User buttons set to monitor the status of the primary also reflect the status of the secondary.
• Depending on the secondary phone type, calls alerting at the secondary but then answered
at the primary may still be logged in the secondary's call log. This occurs if the call log is a
function of the phone rather than the system.

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Configure User Settings

• Call alerting at the secondary phone ignoring any Ring Delay settings applied to the
appearance button being used at the primary phone. The only exception is buttons set to
No Ring, in which case calls are not twinned.
The following applies to internal twinned extensions:
If using a 1400, 1600, 9500 or 9600 Series phone as the secondary extension:
• The secondary extension's directory/contacts functions access the primary user's Centralized
Personal Directory records in addition to the Centralized System Directory.
• The secondary extension's call Log/call List functions access the primary user's Centralized
Call Log.
• The secondary extension's redial function uses the primary users Centralized Call Log. Note
that the list mode or single number mode setting is local to the phone.
It is also shown on 3700 Series phones on a DECT R4 system installed using system
provisioning .
For all phone types, changing the following settings from either the primary or secondary
extension, will apply the setting to the primary user. This applies whether using a short code,
programmable button or phone menu. The status of the function will be indicated on both
extensions if supported by the extension type.
• Forwarding settings.
• Group membership status and group service status.
• Voicemail on/off.
• Do Not Disturb on/off and DND Exceptions Add/Delete.
Mobile Twinning
This method of twinning can be used with external numbers. Calls routed to the secondary remain
under control of the system and can be pulled back to the primary if required. If either leg of an
alerting twinned call is answered, the other leg is ended.
Mobile twinning is only applied to normal calls. It is not applied to:
• Intercom, dial direct and page calls.
• Calls alerting on line appearance, bridged appearance and call coverage buttons.
• Returning held, returning parked, returning transferred and automatic callback calls.
• Follow me calls.
• Forwarded calls except if the user's Forwarded Calls Eligible for Mobile Twinning setting
is enabled.
• Hunt group calls except if the user's Hunt Group Calls Eligible for Mobile Twinning setting
is enabled.
• Additional calls when the primary extension is active on a call or the twinning destination has
a connected twinned call.
A number of controls are available in addition to those on this tab.
Button Programming Actions:
The Emulation | Twinning action can be used to control use of mobile twinning. Set on the
primary extension, when that extension is idle the button can be used to set the twinning

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 Private Calls

destination and to switch twinning usage on/off. When a twinned call has been answered at the
twinned destination, the button can be used to retrieve the call at the primary extension.
Mobile Twinning Handover:
When on a call on the primary extension, pressing the Twinning button will make an unassisted
transfer to the twinning destination. This feature can be used even if the user's Mobile Twinning
setting was not enabled.
• During the transfer process the button will wink.
• Pressing the twinning button again will halt the transfer attempt and reconnect the call at the
primary extension.
• The transfer may return if it cannot connect to the twinning destination or is unanswered
within the user's configured Transfer Return Time (if the user has no Transfer Return Time
configured, a enforced time of 15 seconds is used).
Short Code Features:
The following short code actions are available for use with mobile twinning.
• Set Mobile Twinning Number.
• Set Mobile Twinning On.
• Set Mobile Twinning Off.
• Mobile Twinned Call Pickup.
Related links
Configure User Settings on page 713

Private Calls
This feature allows users to mark a call as being private.
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot
be Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
Use of private calls can be changed during a call. Enabling privacy during a call will stop any
current recording, intrusion or monitoring. Privacy only applies to the speech part of the call. Call
details are still recorded in the SMDR output and other system call status displays.
Button Programming The button programming action Advanced | Call | Private Call can be
used to switch privacy on/off. Unlike the short code features it can be used during a call to apply or
remove privacy from current calls rather than just subsequent calls. On suitable phones the button
indicates the current status of the setting.

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Configure User Settings

Short Codes A number of short code features are available for privacy.
• Private Call Short codes using this feature toggle private status on/off for the user's
subsequent calls.
• Private Call On Short codes using this feature enable privacy for all the user's subsequent
calls until privacy is turn off.
• Private Call Off Short codes using this feature switch off the user's privacy if on.
Related links
Configure User Settings on page 713

System Phone Features

The user option System Phone Rights (User | User) can be used to designate a user as being a
system phone user. System phone users can access a number of additional function not available
to other phone users. Note that if the user has a login code set, they are prompted to enter that
code in order to access these features.
Setting Description
None The user cannot access any system phone options.
Level 1 The user can access all system phone options supported on the type of phone they are
using except system management and memory card commands.
Level 2 The user can access all system phone options supported on the type of phone they
are using including system management and memory card commands. Because of the
nature of the additional commands, you should set a user login code for the user to
restrict access.

System Phone Functions

The following functions are supported:
Feature Description
MENU to set date/ Restricted to 4412, 4424, 6408, 6416 and 6424 phones where supported by the system.
time On these phones, a system phone user can manually set the system date and time by
pressing Menu | Menu | Func | Setup.
Change Login Using a short code with the Change Login Code feature, system phone users can
Code of Other change the login code of other users on the system.
Users
Outgoing Call Bar Using a short code with the Outgoing Call Bar Off feature, system phone users can
Off switch off the outgoing call bar status of other users on the system.

The following commands are only supported using 1400, 1600, 9500, 9600 and J100 Series
phones. Due to the nature of the commands a login code should be set for the user to
restrict access. The commands are accessed through the Features | Phone User | System
Administration menu. For full details refer to the appropriate phone user guide.

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 The 'No User' User

Feature Description
Edit System Using a 1400, 1600, 9500 or 9600 Series phone, a system phone user can edit
Directory Records system directory records stored in the configuration of the system on which they are
hosted. They cannot edit LDAP and/or HTTP imported records.
Date/Time Allows system phone users to manually set the system date and time through a
Programmable programmable button (see System Date and Time on page 663).
Button
The following options are only supported on IP500 V2 systems.
System Allows the user to invoke a system shutdown command.
Management
Memory Card Allows the user to shutdown, startup memory cards and to perform actions to move
Management files on and between memory cards.
System Alarms For certain events the system can display an S on the user's phone to indicate that
there is a system alarm. The user can then view the full alarm text in the phone's
Status menu. The possible alarms in order of priority from the highest first are:
1. Memory Card Failure.
2. Expansion Failure.
3. Voicemail Failure.
4. Voicemail Full.
5. Voicemail Almost Full.
6. License Key Failure.
7. System Boot Error.
8. Corrupt Date/Time.

Related links
Configure User Settings on page 713

The 'No User' User

It is possible to have an extension which has no default associated user. This can occur for a
number of reasons:
• The extension has no Base Extension setting associating it with a user who has the same
setting as their Extension to indicate that they are the extension's default associated user.
• The extension's default associated user has logged in at another extension. Typically they
will be automatically logged back in at their normal extension when they log out the other
phone.
• The extension's default associated user cannot be automatically logged in as they are set to
Forced Login.

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Configure User Settings

Phones with no current user logged in are associated with the setting of the NoUser user in the
system configuration. This user cannot be deleted and their Name and Extension setting cannot
be edited. However their other settings can be edited to configure what functions are available at
extensions with no currently associated user.
By default the NoUser user has Outgoing Call Bar enabled so that the extension cannot be used
for external calls. The users first programmable button is set to the Login action.
Avaya 1100 Series, 1200 Series, M-Series and T-Series phones, when logged out as No User, the
phones are restricted to logging in and dial emergency calls only.
NoUser Source Numbers
The SourceNumbers tab of the NoUser user is used to configure a number of special options.
These are then applied to all users on the system. For details refer to the User | Source Numbers
section.
Related links
Configure User Settings on page 713
Suppressing the NoCallerId alarm on page 730

Suppressing the NoCallerId alarm

Use this procedure to suppress the NoCallerId alarm for all users on the system. Once the task
is completed, the NoCallerID alarm is not raised in SysMonitor, SNMP traps, email notifications,
SysLog or System Status.
Procedure
1. In Manager, in the navigation pane on the left, select User.
2. In the list of users, select NoUser.
3. In the details pane, select the Source Numbers tab.
4. Click Add.
5. In the Source Number field, enter SUPPRESS_ALARM=1.
6. Click OK.
Related links
The 'No User' User on page 729

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Chapter 68: Avaya cloud authorization

Using Avaya cloud authorization, you can configure the Avaya Workplace Client connection using
your Google, Office 365, Salesforce account, Avaya native spaces email account, or Enterprise
Account (SSO).
You can configure the Avaya Workplace Client settings automatically using your email address or
the automatic configuration web address.
Enabling Avaya cloud authorization automatically uses your network login and password to access
different enterprise systems with a single sign-on. Using Avaya cloud authorization, you do not need
to separately login to each system or service in your organization.
For full details, refer to the IP Office SIP Telephone Installation Notes manual.
Note:
Avaya Cloud Account Authorization works only on TLS transport type.
Related links
Apple push notification services on page 731

Apple push notification services

Apple Push Notification service (APNs) is a platform notification service created by Apple Inc.
This service allows iOS users of Avaya Workplace Client for iOS to receive notification of new
calls, voicemail messages, and other events. They receive these notifications regardless when the
Avaya Workplace Client for iOS is idle in the background or is in quit state. However, if Avaya
Workplace Client for iOS is on suspension, then Avaya Workplace Client for iOS automatically
starts when a new call or instant message notification arrives.
Note:
Apple Push Notification service (APNs) works only on TLS transport type.
The iOS device sends notifications via an intermediate push notifications server provided by
Avaya.
Avaya Workplace Client for iOS 3.8 and 3.8.4 supports the push notifications feature.
• On receiving a new call notification, and while Avaya Workplace Client for iOS is on
suspension, it takes up to six seconds before the Avaya Workplace Client for iOS becomes

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Avaya cloud authorization

active, and you can answer the call. The exact delay depends on the version of iOS and
the device used. Therefore, the No Answer Time setting time is increased to more than
20 seconds to allow the calls to ring before going to voicemail or following the divert on no
answer settings.
• APNs service supports only a single iOS device per user. If the you use Avaya Workplace
Client for iOS on two devices, for example, an iPad and an iPhone, only the last client to
register will receive notifications.
• While using iOS push notifications, always configure and enable voicemail or an alternate call
destination number. When Avaya Workplace Client for iOS is not reachable, the No Answer
Time setting triggers and the push notifications to a voicemail or a forward on no answer
number.
• Setting your iOS device with a GSM telephone number as your mobile twinning, and setting
the Mobile Dial Delay (sec) to more than 10 seconds, allows the time for the call notification
to be answered on a previously suspended client before it alerts the GSM call.
Note:
In IP Office,while using iOS push notifications, if you were using a secured port in the primary
server, use the same secured port as a preferred port in secondary server. Any mismatch in
the secured port configuration is not valid.
Related links
Avaya cloud authorization on page 731
Enabling Apple push notifications on page 732

Enabling Apple push notifications

About this task
Apple Push notifications for Avaya Workplace Client on iOS devices.
Use this procedure to enable the push notifications to allow the clients to receive call and
voicemail message notifications.
Note:
When Avaya Workplace Client in your iOS device, such as iPad or iPhone, is in suspended
state or quit state and you log in to another Android or Windows based mobility or desktop
device using the same user, IP Office deletes the associated application device token and
unregisters your iOS device if registered. When you log in using the same user, you must
manually logout and login to the the iOS device to reactivate the token so that you can receive
push notification calls.
Before you begin
• All IP Office’s in Small Community Networking (SCN) should have egress public access to
connect to Apple Push Notification Provider (APNP) to support push notification to Avaya
Workplace Client.
• In the case of SCN deployments,IP Office primary server should synchronize the configured
System ID, Private/Public key with all IP Office in SCN deployments.

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 Apple push notification services

• Configuration synchronize is supported only in IP Office Server Edition with a centralized

primary server (only Star topologies ) and Managed/Hybrid Customer Premises Equipment
(CPE). It is does not in case of Traditional SCN Deployments with 500v2 (includes serial,
mesh and star topologies).
• In the case of Server Edition with centralized primary or Managed/Hybrid CPE, synchronizing
Push details should be done through an explicit button that is available in Web-Manager at
the solution level.
• Synchronizing Push details is dependent on the generation of System-ID (dependent on the
configuration of Zang domain and APNS) and Public Key/Private Key. Enabling APNS at the
solution level, synchronizes the push button.
• Web-Manager needs to synchronize the System-ID and Public/Private key pair in System
Security settings generated in primary server of SCN nodes.
• To sync security settings, the administrator should have access to security settings of IP
Office
• Adding a new expansion to the existing solution synchronizes the configuration to the
expansion. But synchronization of the push details(security settings) should be done
manually using the synchronization button in Web-Manager by Administrator.
• In the case of IP Office 500v2 systems in SCN or Server Edition without a centralized
primary, ensure the company domain is configured and verified.
Procedure
1. Select System Settings > System > Avaya Push Notification Services.
2. Select Enable Apple Push Notification Services.
3. Click OK.

Note:
Increase the No Answer Time settings while using Avaya Workplace Client on iOS
devices to at least 20 seconds. This can be done either by:
• Go to System Settings > System > Telephony > Telephony and increase the
Default No Answer Time settings
• Select Call Management > Users > Add > Telephony > Call Settings and
increase the No Answer Time setting of the individuals.
Related links
Apple push notification services on page 731

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Chapter 69: Managing Users with LDAP

Lightweight Directory Access Protocol (LDAP) is a software protocol for enabling anyone to locate
organizations, individuals, and other resources such as files and devices in a network, whether on
the internet or on a corporate intranet. IP Office supports LDAP version 2 and 3 compliant directory
services servers.
LDAP synchronization allows an administrator to quickly configure the IP Office system with users
and extensions for the users based on an organization’s LDAP directory. An LDAP directory is
organized in a simple tree consisting of the following hierarchy of levels:
1. The root directory (the starting place or the source of the tree)
2. Countries
3. Organizations
4. Organizational units (divisions, departments, etc.)
5. Individuals (which includes people, files, and shared resources, for example printers)
An LDAP directory can be distributed among many servers. Each server can have a replicated
version of the total directory that is synchronized periodically. An LDAP server is called a Directory
System Agent (DSA). An LDAP server that receives a request from a user takes responsibility for
the request, passing it to other DSA's as necessary, but ensuring a single coordinated response for
the user.
Related links
Performing LDAP Synchronization on page 734
Creating a User Provisioning Rule for LDAP Synchronization on page 735

Performing LDAP Synchronization

Procedure
1. In Web Manager, navigate to the page Solution > Solution Settings > User
Synchronization Using LDAP > Connect to Directory Service.
2. Define the connection to the LDAP server and to define the parameters for searching the
LDAP directory. All fields are mandatory.
3. Click Test Connection.
Web Manager attempts to connect to the LDAP server with the specified credentials.

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 Creating a User Provisioning Rule for LDAP Synchronization

4. Click Synchronize User Fields.

5. Map the IP Office user fields to the LDAP fields. Not all fields are mandatory.

Note:
You must click Test Connection on the Connect to Directory Service page to
populate the LDAP fields on the Synchronize User Fields page.
6. Click Preview Results and review the list in the Preview Results window.
7. Click Synchronize.
The User Synchronization window opens. Click the information icon to open a detailed
report.
Related links
Managing Users with LDAP on page 734

Creating a User Provisioning Rule for LDAP

Synchronization
A user provisioning rule (UPR) provides a way to manage the users to be imported. A UPR can
provide the following properties for importing users.
• the IP Office system where the users are created
• starting extension
• extension template
• extension type
• user template
Procedure
1. In Web Manager, navigate to the page Solution > Solution Settings > User
Synchronization Using LDAP > Manage User Provisioning Rules.
2. In the User Provisioning Rule Name field, enter a name for the rule.
3. Optional. Select an IP Office Name from the list.
If an IP Office system is selected, the users are created on this system.
4. Optional. Enter a Start Extension.
If a start extension is provided, users are assigned starting from this extension. If an
extension number is in use, the extension number is skipped and the next available
number is assigned.

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Managing Users with LDAP

Note:
Start Extension is a mandatory field if a value is provided for Extension Template or
Extension Type.
5. Optional. Select an Extension Template from the Select Extension Template list.
The extension template is applied to all users imported with this UPR.
6. Optional. Select an Extension Type to define the extension type created for each user.
If both Select Extension Template and Extension Type are selected, the Extension
Template is used.
7. Optional. Select a User Template from the Select User Template list.
The user template is applied to all users imported with this UPR.
8. In the LDAP directory, enter the name of the UPR created in IP Office in the User column.
9. In IP Office, navigate to the page Solution > Solution Settings > User Synchronization
Using LDAP > Synchronize User Fields.
10. Map the IP Office fields defined in the user provisioning rule to User Provisioning Rule.
Related links
Managing Users with LDAP on page 734

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Chapter 70: Message Waiting Indication

Message waiting indication (MWI) or a message lamp is supported for a wide variety of phones. It is
used to provide the user with indication of when their voicemail mailbox contains new messages. It
can also be configured to provide them with indication when selected hunt group mailboxes contain
new messages.
Avaya digital and IP phones all have in-built message waiting lamps. Also for all phone users, the
one-X Portal for IP Office application provides message waiting indication.
Related links
Message Waiting Indication for Analog Phones on page 737
Message Waiting Indication for Analog Trunks on page 738

Message Waiting Indication for Analog Phones

For analog phones, the system supports a variety of analog message waiting indication (MWI)
methods. The method used for an individual analog extension is set for the Extn | Analog |
Message Waiting Lamp Indication Type field. Those methods are
• 101V
• 51V Stepped
• 81V
• Bellcore FSK
• Line Reversal A
• Line Reversal B
• None
• On
The 101V method is only supported when using a Phone V2 expansion module.
81V is typically used in European countries. 51V Stepped is used in most other countries.
However the actual method used for a particular model of analog phone should be confirmed
with the phone manufacturer's documentation.
The Message Waiting Lamp Indication Type field also provides options for None (no MWI
operation) and On. On selects a default message waiting indication method based on the system
locale.

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Message Waiting Indication

'On' Method Locale

81V Belgium, Denmark, Finland, France, Germany, Greece, Hungary, Iceland, Italy,
Netherlands, Norway, Poland, Portugal, Russia, Saudi Arabia, Sweden, Switzerland,
United Kingdom.
51V Stepped Argentina, Australia, Brazil, Canada, Chile, China, Colombia, Japan, Korea, Mexico,
New Zealand, Peru, South Africa, Spain, United States.

For the United Kingdom system locale (eng), the default Caller Display Type (UK) allows updates
of an analog phone's ICLID display whilst the phone is idle. The system uses this facilities to
display the number of new messages and total number of messages in the users own mailbox.
This feature is not supported with other Caller Display Types.
Hunt Group Message Waiting Indication
By default no message waiting indication is provided for hunt group voicemail mailboxes. Message
waiting indication can be configured by adding an H entry followed by the hunt groups name to
the Source Numbers tab of the user requiring message waiting indication for that hunt group. For
example, for the hunt group Sales, add HSales. Hunt group message waiting indication does not
require the user to be a member of the hunt group.
Related links
Message Waiting Indication on page 737

Message Waiting Indication for Analog Trunks

IP Office can provide a MWI for analog trunks from the PSTN network that terminate on an
ATM4U-V2 card. Multiple users can be configured to receive a MWI from a single analog line.
Users can receive an MWI from multiple lines. Configuring a user for MWI includes configuraton of
a button for automatically dialing the message center.
Note the following conditions.
• Only supported for analog trunks terminating on the ATM4U-V2 card.
• When Analog Trunk MWI is selected as the Voicemail Type, no other voicemail system is
active. As a result, hunt group queue announcements are not supported, since they require
Embedded Voice Mail or Voicemail Pro.
• All analog trunks configured for MWI must use the same message center number. Multiple
message centers are not supported.
• Not supported in One-X Portal.
• No TAPI is provided for analog trunk MWI status.
• Not supported across multiple IP Office systems. If the analog line is on a different node than
the user’s phone, that phone cannot receive an MWI for the line.
• Mobile twinning is not supported. Analog trunk MWI is displayed only on the master set.
• Internal twinning is not supported automatically. However, the twinned set can be configured
to receive the same analog trunk MWI as the master set.

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 Message Waiting Indication for Analog Trunks

Configuring MWI for an Analog Trunk

1. Go to System | Voicemail. In the Voicemail field, select Analog Trunk MWI.
2. In the Destination field, enter the message center telephone number.
3. Select the Line you want to configure for Analog MWI, and then select the Analog
Options tab.
4. In the MWI Standard field, select Bellcore FSK.
5. Select the User you want to configure for MWI and then select the Button Programming
tab.
6. Select the button you want to configure and then click Edit.
7. In the Action field click the browse (...) button and select Advanced > Voicemail >
Monitor Analog Trunk MWI.
8. In the Action Data field, enter the line appearance ID of the analog line.
Related links
Message Waiting Indication on page 737

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Chapter 71: Configuring User Rights

For most settings in a user rights template, the adjacent drop down list is used to indicate whether
the setting is part of the template or not. The drop down options are:
• Apply User Rights Value Apply the value set in the user rights template to all users
associated with the template.

- The matching user setting is grayed out and displays a lock symbol.
- Users attempting to change the settings using short codes receive inaccessible tone.
• Not Part of User Rights Ignore the user rights template setting.
Default User Rights
For defaulted systems, the following user rights are created as a part of the default configuration.
Fields not listed are not part of the user rights.
Note:
When a user logs in as a Outbound Contact Express agent, the Outdialer user rights are
automatically applied. When the agent logs out, the previous user rights are applied.
= Set to On. = Set to Off. – = Not part of the user rights.
User Rights IP Hard Phone Mailbox Paging Outdialer
Priority 5 5 5 5
Voicemail – –
Voicemail Ringback –
Outgoing Call Bar
No Answer Time 0 0 0 0
Transfer Return 0 0 0 0
Time
Individual Coverage 10 10 10 10
Time
Busy on Held – –
Call Waiting
Can Intrude
Cannot be Intruded
Table continues…

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User Rights IP Hard Phone Mailbox Paging Outdialer
Deny Auto Intercom – – –
Calls
Enable Inhibit – – –
Off-Switch Forward/
Transfer
Enable Outgoing – – –
Call Bar
Centralized Logging – – –
Force Login – – –
Force Account Code
Button Programming 1: a= – 1: a=
2: b= 2: b=
3: c= 3: Supervisor
6: Dial *17 4: Extn Logout

User Rights Call Center Agent Boss Application Default

Priority 5 5 5 5
Voicemail – – – –
Voicemail Ringback
Outgoing Call Bar
No Answer Time 0 0 0 0
Transfer Return 0 0 0 0
Time
Individual Coverage 10 10 10 10
Time
Busy on Held
Call Waiting
Can Intrude
Cannot be Intruded
Deny Auto Intercom – – – –
Calls
Enable Inhibit – – – –
Off-Switch Forward/
Transfer
Enable Outgoing – – – –
Call Bar
Centralized Logging – – – –
Force Login – – –
Table continues…

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Configuring User Rights

User Rights Call Center Agent Boss Application Default

Force Account Code
Button Programming 1: a= 1: a= 1: a=
2: b= 2: b= 3: c= 2: b=
4: HGEna 6: DNDOn 3: c=
5: DNDOn 7: Dial *17
6: Busy

Related links
Adding User Rights on page 742
Creating a User Right Based on an Existing User on page 742
Associating User Rights to a User on page 743
Copy User Rights Settings over a User's Settings on page 743

Adding User Rights

Procedure

1. Select User Rights.

2. Click and select User Rights.
3. Enter a name.
4. Configure the user rights as required.
5. Click OK.
Related links
Configuring User Rights on page 740

Creating a User Right Based on an Existing User

About this task
Procedure

1. Select User Rights.

2. In the group pane, right-click and select New User Rights from a User.
3. Select the user and click OK.

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 Associating User Rights to a User

Related links
Configuring User Rights on page 740

Associating User Rights to a User

Procedure

1. Select User Rights or User.

2. In the group pane, right-click and select Apply User Rights to Users.
3. Select the user rights to be applied.
4. On the Members of this User Rights sub tab select the users to which the user rights
should be applied as their Working Hours User Rights.
5. On the Members when out of hours sub tab select which users should use the selected
user rights as their out of hours user rights.
6. Click OK.
Related links
Configuring User Rights on page 740

Copy User Rights Settings over a User's Settings

About this task
This process replaces a user's current settings with those that are part of the selected user rights.
It does not associate the user with the user rights.
Procedure

1. Select User Rights.

2. In the group pane, right-click and select Copy user rights values to users.
3. Select the user rights to be applied.
4. Click OK.
Related links
Configuring User Rights on page 740

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Chapter 72: DND, Follow Me and
Forwarding

This section contains topics looking at how users can have their calls automatically redirected. As
illustrated, there is an order of priority in which the redirect methods are used.
1. Do Not Disturb (DND)
Redirect all calls to voicemail if available, otherwise return busy tone. DND
overrides all the redirect method below unless the calling number is in the
user's DND Exception Numbers List.
2. Follow Me
Redirect all calls to another extension that the users is temporarily sharing.
Follow Me overrides Forward Unconditional. The Follow Me destination is busy
or does not answer, the user's Forward on Busy or Forward on No Answer
options can be used if set.
3. Forward Unconditional
Redirect the user's external calls to another number. That number can be any
number the user can normally dial including external numbers. Forwarding
of hunt group and internal calls is optional. Forward Unconditional overrides
Forward on Busy and Forward on No Answer.
If the destination is an internal user on the same system, they are able to
transfer calls back to the user, overriding the Forward Unconditional.
4. Forward on Busy
Redirects the user's external calls when the system sees the user as being
busy. Uses the same number as Forward Unconditional unless a separate
Forward on Busy Number is set. Forwarding internal calls is optional. Forward
on Busy overrides Forward on No Answer.
5. Forward on No Answer
Redirects the user's external calls when they ring for longer than the user's
No Answer Time. Uses the same number as Forward Unconditional unless a
separate Forward on Busy Number is set. Forwarding internal calls is optional.

Retrieving Externally Forwarded Calls:

Where a call is forwarded to an external destination and receives busy or is not answered within the
forwarding user's No Answer Time, the system will attempt to retrieve the call. If forwarded on a
trunk that does not indicate its state the call is assumed to have been answered, for example analog
loop start trunks.

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 Do Not Disturb (DND)

Off-Switch Forwarding Restrictions:

User forwarding is subject to the same restrictions as transferring calls. To bar a user from
forwarding calls to an external number, the Inhibit Off-Switch Forward/Transfers (User |
Telephony | Supervisor Settings) option. To bar all users from forwarding calls to external
numbers the Inhibit Off-Switch Forward/Transfers option can be used.
When transferring a call to another extension that has forwarding enabled, the type of call being
transferred is used. For example, if transferring an external call, if the transfer target has forwarding
of external calls enabled then the forward is used.
Block Forwarding:
The Block Forwarding setting is used for enforcing predictable call routing, where the call should
always go to the same destination. This setting was implemented for contact center applications.
Block Forwarding can be set for a user on the User | Forwarding page or as a user rights setting on
the User Rights | Forwarding page.
Related links
Do Not Disturb (DND) on page 745
Follow Me on page 747
Forward Unconditional on page 749
Forward on Busy on page 751
Forward on No Answer on page 753
Determining a User's Busy Status on page 755
Chaining on page 756

Do Not Disturb (DND)

Summary: Redirect all calls to busy tone or to voicemail if available except those in your DND
exceptions list.
Do Not Disturb (DND) is intended for use when the user is present but for some reason does not
want to be interrupted. Instead calls are sent to voicemail if available, otherwise they receive busy
tone.
• Exceptions Specific numbers can be added to the user's Do Not Disturb Exception List.
Calls from those numbers override DND. N and X wildcards can be used at the end of
exception numbers to match a range of numbers. For external numbers, this uses the
incoming caller line ID (ICLID) received with the call.
• Priority Enabling DND overrides any Follow Me or forwarding set for the user, except for
calls in the user's Do Not Disturb Exception List.
• Phone When enabled, the phone can still be used to make calls. An N is displayed on
many Avaya phones. When a user has do not disturb in use, their normal extension will give
alternate dialtone when off hook.
Applied to

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DND, Follow Me and Forwarding

Call Types Blocked Call Treatment

Internal Voicemail if available, otherwise busy tone.
External Voicemail if available, otherwise busy tone.
Hunt Group Call not presented (DND exceptions are not used).
Page Call not presented.
Follow Me Rings.
Forwarded Busy.
VM Ringback Rings
Automatic Callback Rings
Transfer Return Rings.
Hold Return Rings.
Park Return Rings.
Twinning Voicemail if available, otherwise busy tone.

Do Not Disturb and Twinning

• Mobile Twinning Selecting DND disables mobile twinning.
• Internal Twinning
- Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.
- Logging out or setting do not disturb at the secondary only affects the secondary.
• Do Not Disturb Exceptions List For both types of twinning, when DND is selected, calls
from numbers entered in the user's Do Not Disturb Exception List are presented to both the
primary and secondary phones.
Do Not Disturb Controls
Do Not Disturb
Manager A user's DND settings can be viewed and changed through the User | DND tab within the
system configuration settings.
Controls The following short code features/button programming actions can be used:

Voicemail If voicemail is available, it is used instead of busy tone for callers not in the users
exceptions list.
For Voicemail Pro, the Play Configuration Menu action can be used to let callers switch
DND on or off.
SoftConsole A SoftConsole user can view and edit a user's DND settings except exception numbers.
Through the directory, select the required user. Their current status including DND is
shown. Double-click on the details to adjust DND on or off.

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 Follow Me

Feature/Action Short Code Default Button

Do Not Disturb On *08 - Toggles.
Do Not Disturb Off *09
Do Not Disturb Exception *10*N#
Add
Do Not Disturb Exception *11*N#
Delete
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 744

Follow Me
Summary: Have your calls redirected to another user's extension, but use your coverage,
forwarding and voicemail settings if the call receives busy tone or is not answered.
Follow Me is intended for use when a user is present to answer calls but for some reason is
working at another extension such as temporarily sitting at a colleague's desk or in another office
or meeting room. Typically you would use Follow Me if you don't have a Hot Desking log in code or
if you don't want to interrupt your colleague from also receiving their own calls, ie. multiple users at
one phone.
• Priority
Follow Me is overridden by DND except for callers in the user's DND Exception Numbers
List. Follow Me overrides Forward Unconditional but can be followed by the user's Forward
on Busy or Forward on No Answer based on the status of the Follow Me destination.
• Destination
The destination must be an internal user extension number. It cannot be a hunt group
extension number or an external number.
• Duration
The Follow Me user's no answer timeout is used. If this expires, the call either follows their
Forward on No Answer setting if applicable, or goes to voicemail is available. Otherwise the
call continues to ring at the destination.
• Phone
When enabled, the phone can still be used to make calls. When a user has follow me in use,
their normal extension will give alternate dial tone when off hook.
• Exceptions
- The Follow Me destination extension can make and transfer calls to the follow me source.

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DND, Follow Me and Forwarding

- The call coverage settings of the user are applied to their Follow Me calls. The call
coverage settings of the destination are not applied to Follow Me calls it receives.
Call Types Redirected
Internal Redirected.
External Redirected.
Hunt Group Redirected*.
Page Redirected.
Follow Me Not redirected.
Forwarded Redirected.
VM Ringback Not redirected.
Automatic Callback Not redirected.
Transfer Return Not redirected.
Hold Return Not redirected.
Park Return Not redirected.

*Except calls for "Longest Waiting" type hunt groups.

Follow Me Controls
Manager A user's Follow Me settings can be viewed and changed through the User | Forwarding
tab within the system configuration settings. Note that on this tab, entering a Follow Me
Number also enables Follow Me.
Controls The following short code features/button programming actions can be used:

Voicemail For calls initially targeted to the user but then redirected, when voicemail is invoked the
mailbox of the user is used and not the mailbox of the destination.
For Voicemail Pro, the Play Configuration Menu action can be used to let callers alter or
set their current Follow Me destination.
SoftConsole A SoftConsole user can view and edit a user's Follow Me settings. Through the directory,
select the required user. Their current status including Follow Me is shown. Double-click
on the details and select Forwarding to alter their forwarding settings including Follow
Me.

Feature/Action Short Code Default Button

Follow Me Here *12*N#
Follow Me Here Cancel *13*N#
Follow Me To *14*N#
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 744

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 Forward Unconditional

Forward Unconditional
Summary: Have your calls redirected immediately to another number including any external
number that you can dial.
• Priority
This function is overridden by DND and or Follow Me if applied. Forward Unconditional
overrides Forward on Busy.
• Destination
The destination can be any number that the user can dial. If external and Inhibit Off-Switch
Transfers is applied, the caller is directed to voicemail if available, otherwise they receive
busy tone. If the destination is an internal user on the same system, they are able to transfer
calls back to the user, overriding the Forward Unconditional.
• Duration
After being forwarded for the user’s no answer time, if still unanswered, the system can apply
additional options. It does this if the user has forward on no answer set for the call type or if
the user has voicemail enabled.
- If the user has forward on no answer set for the call type, the call is recalled and then
forwarded to the forward on no answer destination.
- If the user has voicemail enabled, the call is redirected to voicemail.
- If the user has both options set, the call is recalled and then forwarded to the forward on
no answer destination for their no answer time and then if still unanswered, redirected to
voicemail.
- If the user has neither option set, the call remains redirected by the forward unconditional
settings.
Note that for calls redirected via external trunks, detecting if the call is still unanswered requires
call progress indication. For example, analog lines do not provide call progress signalling and
therefore calls forwarded via an analog lines are treated as answered and not recalled.
• Phone
When enabled, the phone can still be used to make calls. An D is displayed on DS phones.
When a user has forward unconditional in use, their normal extension will give alternate
dialtone when off hook.
• Calls Forwarded
Once a call has been forwarded to an internal destination, it will ignore any further Forward
No Answer or Forward on Busy settings of the destination but may follow additional
Forward Unconditional settings unless that creates a loop.
Call Types Forwarded
Internal Optional.
Table continues…

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DND, Follow Me and Forwarding

Call Types Forwarded

External Forwarded.
Hunt Group Optional.*
Page Not presented.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

*Optional only for calls targeting sequential and rotary type groups. Includes internal call to a hunt
group regardless of the forward internal setting.
• To Voicemail: Default = Off.
If selected and forward unconditional is enabled, calls are forwarded to the user's voicemail
mailbox. The Forward Number and Forward Hunt Group Calls settings are not used. This
option is not available if the system's Voicemail Type is set to None. 1400, 1600, 9500 and
9600 Series phone users can select this setting through the phone menu. Note that if the
user disables forward unconditional the To Voicemail setting is cleared.
Forward Unconditional Controls
Forward Unconditional Controls
Manager A user's forwarding settings can be viewed and changed through the User | Forwarding
tab within the system configuration settings.
Controls The following short code features/button programming actions can be used:

Voicemail For calls initially targeted to the user but then redirected, when voicemail is invoked the
mailbox of the user is used and not the mailbox of the destination.
For Voicemail Pro, the Play Configuration Menu action can be used to let callers set their
current forwarding destination and switch Forwarding Unconditional on/off.
SoftConsole A SoftConsole user can view and edit a user's forwarding settings. Through the directory,
select the required user. Their current forwarding status is shown. Double-click on the
details and select Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward Unconditional On *01 - Toggles.
Forward Unconditional Off *02
Table continues…

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 Forward on Busy

Feature/Action Short Code Default Button

Forward Hunt Group Calls - Toggles.
On
Forward Hunt Group Calls
Off
Disable Internal Forwards
Enable Internal Forwards
Disable Internal Forwards
Unconditional
Enable Internal Forwards
Unconditional
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 744

Forward on Busy
Summary: Have your calls redirected when you are busy to another number including any
external number that you can dial.
The method by which the system determines if a user is 'busy' to calls depends on factors such as
whether they have multiple calls appearance buttons or Call Waiting and or Busy on Held set. See
Busy.
• Priority
This function is overridden by DND and or Forward Unconditional if applied. It can be applied
after a Follow Me attempt. It overrides Forward on No Answer.
• Destination
The destination can be any number that the user can dial. The Forward Unconditional
destination number is used unless a separate number Forward on Busy Number is set.
If Inhibit Off-Switch Transfers is applied, the caller is directed to voicemail if available,
otherwise they receive busy tone.
• Duration
The destination is rung using the forwarding user's No Answer Time. If this expires, the call
goes to voicemail is available. Calls to an external destination sent on trunks that do not
signal their state are assumed to have been answered, for example analog loop start trunks.
• Phone
Forward on Busy is not indicated and normal dial tone is used.

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DND, Follow Me and Forwarding

• Calls Forwarded
Once a call has been forwarded to an internal destination, it will ignore any further Forward
No Answer or Forward on Busy settings but may follow additional Forward Unconditional
settings.
Call Types Forwarded
Internal Optional.
External Forwarded.
Hunt Group Not presented.
Page Not presented.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

Forward on Busy Controls

Software Level A user's forwarding settings can be viewed and changed through the User | Forwarding
tab within the system configuration settings.
Controls The following short code features/button programming actions can be used:

Voicemail For calls initially targeted to the user but then redirected, when voicemail is invoked the
mailbox of the user is used and not the mailbox of the destination.
For Voicemail Pro, the Play Configuration Menu action can be used to let callers set the
forward destination.
SoftConsole A SoftConsole user can view and edit a user's forwarding settings. Through the directory,
select the required user. Their current forwarding status is shown. Double-click on the
details and select Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward on Busy Number *57*N#
Forward on Busy On *03 - Toggles.
Forward on Busy Off *04
Disable Internal Forwards
Enable Internal Forwards
Disable Internal Forwards
Busy or No Answer
Table continues…

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 Forward on No Answer

Feature/Action Short Code Default Button

Enable Internal Forwards
Busy or No Answer
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 744

Forward on No Answer
Summary: Have your calls redirected another number if it rings without being answered.
• Priority
This function is overridden by DND and Forward on Busy if applied. It can be applied after
a Follow Me attempt. Forward Unconditional overrides Forward on Busy and Forward on No
Answer.
• Destination
The destination can be any number that the user can dial. The Forward Unconditional
destination number is used unless a separate number Forward on Busy Number is set.
If Inhibit Off-Switch Transfers is applied, the caller is directed to voicemail if available,
otherwise they receive busy tone.
• Duration
The destination is rung using the forwarding user's No Answer Time. If this expires, the call
goes to voicemail is available. Otherwise the call continues to ring at the destination. Calls to
an external destination sent on trunks that do not signal their state are assumed to have been
answered, for example analog loop start trunks.
• Phone
Forward on No Answer is not indicated and normal dial tone is used.
• Calls Forwarded
Once a call has been forwarded to an internal destination, it will ignore any further Forward
No Answer or Forward on Busy settings but may follow additional Forward Unconditional
settings.
Call Types Forwarded
Internal Optional.
External Forwarded.
Hunt Group Not applicable.
Table continues…

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DND, Follow Me and Forwarding

Call Types Forwarded

Page Not applicable.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

Forward on No Answer Controls

Manager A user's forwarding settings can be viewed and changed through the User | Forwarding
tab within the system configuration settings.
Controls The following short code features/button programming actions can be used:

Voicemail For calls initially targeted to the user but then redirected, when voicemail is invoked the
mailbox of the user is used and not the mailbox of the destination.
For Voicemail Pro, the Play Configuration Menu action can be used to let callers set
the forward destination. It cannot however be used to enable Forward on Busy or set a
separate Forward on Busy number.
SoftConsole A SoftConsole user can view and edit a user's forwarding settings. Through the directory,
select the required user. Their current forwarding status is shown. Double-click on the
details and select Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward on Busy Number *57*N#
Forward on No Answer *05 - Toggles.
On
Forward on No Answer *06
Off
Enable Internal Forwards
Disable Internal Forwards
Enable Internal Forwards
Busy or No Answer
Disable Internal Forwards
Busy or No Answer
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 744

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 Determining a User's Busy Status

Determining a User's Busy Status

Various system features allow users to handle more than one call at a time. Therefore the term
"busy" has different meanings. To other users it means whether the user is indicated as being
busy. To the system it means whether the user is not able to receive any further calls. The latter
is used to trigger 'busy treatment', either using a user's Forward on Busy settings or redirecting
calls to voicemail or just returning busy tone.
• Busy Indication - In Use
The user busy indication provided to programmable buttons and to user applications, is
based on the monitored user's hook switch status. Whenever the user is off-hook, they will be
indicated as being busy regardless of call waiting or call appearance settings.
• Busy to Further Calls
Whether a user can receive further calls is based on a number of factors as described below.
- Logged In and Present
Is the user logged into an extension and is that extension physically connected to the
system.
- Busy on Held
If a user enables their Busy on Held setting, whenever they have a call on hold, they are
no longer available to any further incoming calls.
- Appearance Buttons
A user's call appearance button are used to receive incoming calls. Normally, whilst the
user has any free call appearance buttons, they are available to receive further calls.
Exceptions are:
• Reserve Last Appearance
Users with appearance buttons require a free call appearance button to initiate transfers
or conferences. Therefore it is possible through the user's configuration settings to
reserve their last call appearance button for outgoing calls only.
• Other Appearance Buttons
Calls may also be indicated on line, call coverage and bridged appearance buttons.
• Call Waiting
Users of phones without appearance buttons can use call waiting. This adds an audio tone,
based on the system locale, when an additional call is waiting to be answered. Only one
waiting call is supported, any further calls receive busy treatment.
• Hunt Group Calls
A user's availability to receive hunt group calls is subject to a range of other factors. See
Member Availability.
Related links
DND, Follow Me and Forwarding on page 744

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DND, Follow Me and Forwarding

Chaining
Chaining is the process where a call forward to an internal user destination is further forwarded by
that user's own forwarding settings.
• Follow Me Calls
Follow Me calls are not chained. They ignore the forwarding, Follow Me and Do Not Disturb
settings of the Follow Me destination.
• Voicemail
If the call goes to voicemail, the mailbox of the initial call destination before forwarding is
used.
• Looping
When a loop would be created by a forwarding chain, the last forward is not applied. For
example the following are scenarios where A forwards to B, B forwards to C and C forwards
to A. In each case the final forward is not used as the destination is already in the forwarding
chain.

• Hunt Group Loop

If a user forwards a call to a hunt group of which they are a member, the group call is not
presented to them but is presented to other members of the hunt group.
• Maximum Number of Forwards
A maximum of 10 forwarding hops are supported for any call.
• Calls Forwarded
Once a call has been forwarded to an internal destination, it will ignore any further Forward
No Answer or Forward on Busy settings but may follow additional Forward Unconditional
settings.

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 Chaining

Related links
DND, Follow Me and Forwarding on page 744

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Chapter 73: Hot Desking

Hot desking allows users to log in at another phone. Their incoming calls are rerouted to that phone
and their user settings are applied to that phone. There are a number of setting and features which
affect logging in and out of system phones.
To hot desk, a user must be assigned a Login Code (User > Telephony > Supervisor Settings) in
the system configuration.
By default, each system extension has an Base Extension setting. This associates the extension
with the user who has the matching Extension settings as being that extension's default associated
user.
• By leaving the Base Extension setting for an extension blank, it is possible to have an
extension with no default associated user. This is only supported for non-IP/CTI extensions.
Extensions in this state use the settings of a special user named NoUser. On suitable phones
the display may show NoUser.
• You can create users whose Extension directory number is not associated with any physical
extension. These users must have a log in code in order to log in at a phone when they need
to make or receive calls. In this way the system can support more users than it has physical
extensions.
• Remote extensions must have an associated default user who is logged in. That user's user
profile establishes the extension's right to operate as a remote extension. Any other user
logging in over the default user must also have a user profile that allows remote extension
usage.
Related links
Hot Desking Operation on page 759
Logging Out on page 759
Hot Desking Controls on page 760
Hot Desking in an IP Office Network on page 760
Call Center Agents on page 761
Hot Desking Examples on page 761
Automatic Log Out on page 763

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 Hot Desking Operation

Hot Desking Operation

When another user logs in at an extension, they control that phone. Any existing user, including
the default associated user, is logged out of that phone.
• Any user settings not applicable to the type of phone on which the user has logged
in become inaccessible. For example some programmable button features will become
inaccessible if the phone at which a user logs in does not have a sufficient programmable
buttons.
• 1400 Series, 1600 Series, 9500 Series, 9600 Series and J100 Series telephones all use the
centralized call log and centralized personal directory features that move those settings with
the user as they hot desk.
• Other Avaya H.323 IP telephones can be configured to backup and restore user settings to a
file server when a user hot desks between phones. The range of settings supported depends
on the particular phone model. Refer to the Avaya IP Office™ Platform H.323 Telephone
Installation manual.
• For all other features and phone types, it must be assumed that any settings and data shown
by the phone is stored by the phone and are still accessible after logging off.
• By default the IP Office system blocks J129 and H175 phones from being used for
hot-desking. If required, the NoUser source number SIP_ENABLE_HOT_DESK enables hot-
desking support for those phones.
• Hot-desking is not supported for SIP softphone applications. That includes clients running on
Avaya Vantage™ telephones.
Related links
Hot Desking on page 758

Logging Out
When a user logs out or is logged out by someone else logging in, they are automatically logged
back in at the extension for which they are the default associated user if no one else is logged
in at that extension. However this does not happen for users set to Forced Login (User >
Telephony > Supervisor Settings).
• For each user, you can configure how long the extension at which they are logged in can
remain idle before they are automatically logged out. This is done using the Login Idle Period
option. This option should only be used in conjunction with Force Login.
• Logged in users who are members of a hunt group can be automatically logged out if they do
not answer hunt group calls presented to them. This is done by selecting Logged Off as the
user's Status on No Answer (User > Telephony > Supervisor Settings) setting.
• Calls to a logged out user are treated as if the user is busy until the user logs in.

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Hot Desking

Related links
Hot Desking on page 758

Hot Desking Controls

Logging in and out at a phone can be done either using system short codes or programmable
buttons.
• The default system short code for logging in, is *35*N# where the user replaces N with
their extension number and log in code separated by a *. This uses the short code feature
ExtnLogin. If the user dials just a log in code as N, it is checked against the user with the
same extension number as the extension's base extension number.
• The default system short code for logging out is *36. This uses the short code feature
ExtLogout.
• The ExtnLogin and ExtnLogout features can be assigned to programmable buttons on
suitable Avaya phones. The ExtnLogin button prompts the user to enter their details.
Related links
Hot Desking on page 758

Hot Desking in an IP Office Network

Hot desking can be used in a network of IP Office systems.
• The IP Office system on which the user is configured is termed their 'home' system
• All other IP Office systems are 'remote' systems.
The following additional features are supported for hot-desking with a network of IP Office
systems.
Hot Desking onto another IP Office System
The system supports hot desking between systems within a network of IP Office systems. In the
descriptions below:
When a user logs in to a remote system:
• The user's incoming calls are automatically rerouted to the remote IP Office system.
• The user's outgoing calls uses the settings of the remote IP Office system.
• The user's license privileges move with them. For example, their user profile setting is
retained with the remote IP Office needing licenses for that profile type.

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 Call Center Agents

• The user's own settings are transferred. However, some settings may become unusable or
may operate differently:
- User rights are not transferred to the remote system but the name of any user rights
associated with the user are transferred. If user rights with the same name exist on the
remote system, then they are used. The same applies for user rights applied by time
profiles, if time profiles with the same name exist on the remote system .
- Appearance buttons configured for users on the home system will no longer operate.
- Various other settings may either no longer work or may work differently depending on the
configuration of the remote system at which the user has logged in.
If the user's home system is disconnected from the network while the user is remotely hot desked,
the user remains remotely hot desked. They can remain in that state unless the remote system
is restarted. Note however, when the user's home system is reconnected, the user may be
automatically logged back onto that system.
Dialing from another IP Office System (Break Out)
In some scenarios a hot desking user logged in at a remote system will want to dial a number
using the system short codes of another system, typically their home system. This can be done
using either short codes with the Break Out feature or a programmable button set to Break Out.
This feature can be used by any user within the multi-site network but is of most use to remote hot
desked users.
Related links
Hot Desking on page 758

Call Center Agents

On systems with a call center application such as Compact Contact Center (CCC) or Compact
Business Center (CBC), logging in and logging out is a key part of tracking and reporting on
call center agents. It also controls call distribution as, until the agent logs in, their hunt group
membership is seen as disabled.
For CCC, CBC and Delta Server, an agent is defined as being a user with a Login Code and set to
Forced Login. Those users consume a CCC agent license.
Related links
Hot Desking on page 758

Hot Desking Examples

The following are example of different ways that the hot desking settings can be used.
Related links
Hot Desking on page 758

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Hot Desking

Scenario 1: Occasional Hot Desking

About this task
In this scenario, a particular user, for this example extension 204, needs to occasionally work at
other locations within the building.
Procedure
1. A Login Code is added to the user's configuration settings, for this example 1234.
2. The user can now log in when needed at any other phone by dialing *35*204*1234#.
The phone's default associated user is logged out by this and their calls get busy
treatment. User 204 is also logged out their normal phone and their calls now rerouted
to the phone at which they have logged in.
3. When finished, the user can dial *36 to log out.
4. This logs the phone's normal default user back on.
Its also logs the hot desking user back on at their normal extension.

Scenario 2: Regular Hot Desking

About this task
This scenario is very similar to the one above. However, the user doesn't want to be automatically
logged back in on their normal phone until they return to its location.
Procedure
1. A Login Code is added to the user's configuration settings, for this example 1234.
2. The Forced Login option is selected.
3. When the user logs out of the phone that they are currently using, they are no longer
automatically logged in on their normal extension.
When they return to it they must dial *35*204*1234# to log in.
4. Whilst not logged in anywhere, calls to the user receive busy treatment.

Scenario 3: Full Hot Desking

About this task
Similar to the scenarios above but this time the user doesn't have a regular phone extension that
they use. In order to make and receive calls they must find a phone at which they can log in.
Procedure
1. The user is given an Extension directory number that is not matched by the extension
directory number setting of any existing extension.

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 Automatic Log Out

2. They are also given a Login Code and a Login Idle Period is set, for this example 3600
seconds (an hour). Forced Login isn't required as the user has no default extension at
which they might be automatically logged in by the system.
3. The user can now log in at any available phone when needed.
4. If at the end of the business day they forget to log out, the Login Idle Period will eventually
log them off automatically.

Scenario 4: Call Center Hot Desking

About this task
In this scenario, the phone extensions have no default extension number. Several phones set
like this might be used in a call center where the agents use whichever desk is available at the
start of their shift. Alternatively a set of desks with such phones might be provided for staff that
are normally on the road but occasionally return to the office and need a temporary desk area to
complete paper work.
Procedure
1. For the extensions, the Extension setting is left blank.
This means that those phones will be associated with the NoUser user's settings and
display NOT LOGGED ON.
2. The call center agents or road-warrior users are configured with Extension directory
numbers that also don't match any existing physical extensions.
They are all given Login Code numbers.
3. The users can log in at any of the extensions when required.
When they log out or log in elsewhere, the extensions return to the NoUser setting.

Automatic Log Out

Normally a user can either log themselves out or be logged out by another user logging in. The
following methods can be used by the system to automatically log out a user, so long as that user
has a Login Code and is set to Forced Login.
Note: A remote hot desking user whose home system can no longer be seen by the remote
system at which they are logged in is automatically logged out after 24 hours.
Idle Timeout:
The user Login Idle Period (User | Telephony | Supervisor Settings) can be used to
automatically log out the user after a set period of phone inactivity. The period can be set between
1 to 99999 seconds and is based on call inactivity other than ringing calls.

Unanswered Calls:

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Hot Desking

Users who are members of hunt groups are presented with hunt group calls when they are logged
in and not already on a call. If the user is logged in but not actually present they will continue to be
presented with hunt group calls. In this scenario it can be useful to log the user off.
• For the hunt group On the Hunt Group | Hunt Group tab, use the Agent's Status on
No Answer Applies to setting to select which types of unanswered hunt group calls should
change the user's status. The options are:
- None
- Any Calls
- External Inbound Calls Only
• For the user The Status on No Answer setting (User | Telephony | Supervisor Settings)
can be used. This sets what the user's status should be changed to if they do not answer a
hunt group call. The options are:
- Logged In If this option is selected, the user's status is not changed.
- Busy Wrap-Up If this option is selected, the user's membership status of the hunt group
triggering the action is changed to disabled. The user can still make and receive calls and
will still continue to receive calls from other hunt groups to which they belong.
- Busy Not Available If this option is selected, the user's status is changed to do not
disturb. This is the equivalent of DND and will affect all calls to the user.
- Logged Off If this option is selected, the user’s status is changed to logged out. In that
state the cannot make calls and cannot receive calls. Hunt group calls go to the next
available agent and personal calls treat the user as being busy.
Related links
Hot Desking on page 758

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Chapter 74: Group Operation

A group is a collection of users accessible through a single directory number. Calls to that group
can be answered by any available member of the group. The order in which calls are presented can
be adjusted by selecting different group types and adjusting the order in which group members are
listed.

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Group Operation

Fallback
checks

Availability
checks

Call
presentation

Queuing

Overflow

Voicemail

• Call Presentation: The order in which the available members of the group are used for call

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 presentation is selectable.
• Availability: There are a range of factors which control whether group calls are presented to a
user in addition to that user being a member of the group.
• Queuing: This optional feature allows calls to be queued when the number of calls to be
presented exceeds the number of available group members to which call can be presented.
• Announcements: On systems with a voicemail server (Voicemail Pro or Embedded
Voicemail), announcements can be played to callers waiting to be answered. That includes
calls that are ringing and calls that are queued.
• Overflow: This optional feature can be used to include additional agents from an overflow
group or groups when a call is not answered.
• Fallback: A group can be taken out of operation manually or using a time profile. During
fallback, calls can be redirected to a fallback group or sent to voicemail or just receive busy
tone. Two types of fallback are supported; night service and out of service.
• Voicemail: Calls can be redirected to voicemail. The system allows selection of whether group
calls remain in the group mailbox or are copied (broadcast) to the individual mailboxes of the
group members. When messages are stored in the group's own mailbox, selection of who
receives message waiting indication is possible.
Group Editing
Changing the name of a group has the following effects:
• A new empty mailbox is created on voicemail with the new group name.
• Records in other groups' Overflow lists will be updated.
• Out-of-Service and Night-Service fallback references are updated.
Modifying the extension number of a group updates the following:
• Group buttons.
• Overflow, Out of Service Fallback and Night Service Fallback group records.
• Incoming call route records.
When a group is deleted, all references to the deleted group will be removed including:
• Records in Incoming call routing tables.
• Transfer target in internal auto-attendant.
• Overflow, Night-Service or Fallback-Service on other groups.
• DSS keys monitoring group status.
Server Edition Group Management
Groups can be stored in the configuration of any system in the network. Groups created at the
solution level on Manager and Web Manager are stored on the Primary Server. All groups can
include users from anywhere in the network and are automatically advertised to and diallable on any
of the systems in the network.
Groups configured on the Server Edition Primary by default fail over to the Server Edition
Secondary. Groups configured on a Server Edition Expansion System can be configured to fail over
to the Server Edition Primary, the Server Edition Secondary, or another Server Edition Expansion
System.

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Group Operation

Groups in a Multi-Site Network

In a multi-site network, the extension numbers of users are automatically shared between systems
and become diallable from other systems without any further programming.
The following features are available for groups.
Advertised Groups:
Each group can be set as being 'advertised'. The group can then be dialed from other systems
within the multi-site network. The groups extension number and name must be unique within the
network. Non-advertised group numbers remain local only to system hosting the group.
Distributed Groups:
Groups on a system can include users located on other systems within the network. Distributed
groups are automatically advertised to other systems within the network. Note that distributed
groups can only be edited on the system on which they were created.
Related links
Group Types on page 768
Call Presentation on page 769
Group Member Availability on page 771
Example Hunt Group on page 773
CBC/CCC Agents and Hunt Groups on page 774
Coverage Groups on page 775

Group Types
At its most basic, a group’s settings consist of a group name, an extension number, a list of group
members and a hunt type selection. It is the last two settings which determine the order in which
incoming calls are presented to hunt group members.
The available group types are; Collective, Sequential, Rotary and Longest Waiting. These work
are follows:
Collective Group
An incoming call is presented simultaneously to all the available group members.

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 Call Presentation

Sequential Group
An incoming call is presented to the first available member in the list. If unanswered, it is
presented to the next available member in the list.
The next incoming call uses the same order. It is presented to the available members starting
again from the top of the list.

Rotary Hunt Type

This hunt type operates similarly to Sequential. However the starting point for call presentation is
the first available member after the last member to answer a call.

Longest Waiting Hunt Type

Where hunt group calls are being presented to a twinned extension, the longest waiting status of
the user can be reset by calls answered at either their master or twinned extension.
An incoming call is first presented to the available member who has been idle the longest. If
unanswered it is presented to the next longest idle member.
This hunt type does not present calls to hunt group members in the order that they are listed. It
presents calls using the order of how long the available hunt group members have been idle.
Related links
Group Operation on page 765

Call Presentation
Summary: Calls are presented to each available hunt group member in turn. If having been
presented to all the available members, none answers, the call is redirected to voicemail if
available, otherwise it continues to be presented to the next available member.

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Group Operation

In addition to the summary, options exist to have calls queued or to have calls also presented to
agents in an overflow group or groups.
• First and Next Available Members
The first available member to which a call is presented and the order of the next available
members to which a call is presented are determined by the hunt group's Hunt Type setting.
• Additional Calls
When additional calls are waiting to be presented, additional available hunt group members
are alerted using the hunt group type. When any member answers a call it will be the first
waiting call that is answered.
• No Available Members
If the number of incoming calls exceeds the number of available members to which calls can
be presented, the following actions are usable in order of precedence.
• Queuing
If queuing has been enabled for the hunt, it is applied to the excess calls up to the limits
specified for the number of queued calls or length of time queued.
• Voicemail
If voicemail has been enabled for the hunt group, excess calls are directed to voicemail.
• Busy Tone
Busy tone is returned to the excess calls (except analog and T1 CAS calls which remain
queued).
• No Answer Time
This value is used to determine how long a call should ring at a hunt group member before
being presented to the next available hunt group member. The System | Telephony |
Telephony | No Answer Time setting is used unless a specific Hunt | Hunt Group | No
Answer Time is set.
• Voicemail
If voicemail is being used, if having been presented to all the available group members the
call is still not answered then it goes to voicemail.
- The call will also go to voicemail when the hunt group's Voicemail Answer Time is
exceeded. the mailbox of the originally targeted hunt group is used even if the call has
overflowed or gone to a night server hunt group.
• Calls Not Being Answered Quick Enough - Overflow
In addition to ringing at each available member for the No Answer Time, a separate Overflow
Time can be set. When a calls total ring time against the group exceeds this, the call can be
redirected to an overflow group or groups.
• No Available Member Answers
If a call has been presented unanswered to all the available members, either of two actions
can be applied. If voicemail is available, the call is redirected to voicemail. If otherwise, the

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 Group Member Availability

call will continue being presented to hunt group members until answered or, if set, overflow is
used.
• Call Waiting
For hunt groups using the Group hunt type, call waiting can be used.
Related links
Group Operation on page 765

Group Member Availability

Summary: Details when a hunt group member is seen as being available to be presented a hunt
group call.
The Hunt Group settings within Manager list those users who are members of the hunt group and
therefore may receive calls directed to that hunt group. However there are a range of factors that
can affect whether a particular hunt group member is available to take hunt group calls at any
time.
• Existing Connected Call
Users with an existing connected call are not available to further hunt group calls. This is
regardless of the type of connected call, whether the user has available call appearance
buttons or is using call waiting.
• Hunt Group Call Waiting
For Collective hunt groups call waiting can be enabled using the Ring Type of Collective
Call Waiting.
• Logged In/Logged Out
The system allows user's to log in and out extensions, a process known as 'hot desking'.
Whilst a user is logged out they are not available to receive hunt group calls.
- Mobile Twinning users with both Hunt group calls eligible for mobile twinning and Twin
when logged out selected will still receive hunt group calls unless they switch off twinning.
• Membership Enabled/Disabled
The system provides controls to temporarily disable a users' membership of a hunt group.
Whilst disabled, the user is not available to receive calls directed to that hunt group.
• Do Not Disturb
This function is used by users to indicate that they do not want to receive any calls. This
includes hunt group calls. In call center environments this state is also known as 'Busy Not
Available'. See Do Not Disturb.
• Busy on Held

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Group Operation

When a user has a held call, they can receive other calls including hunt group calls. The
Busy on Held settings can be used to indicate that the user is not available to further calls
when they have a held call.
• Forward Unconditional
Users set to Forward Unconditional are by default not available to hunt group calls. The
system allows the forwarding of hunt group calls to be selected as an option.
• Idle /Off Hook
The hunt group member must be idle in order to receive hunt group call ringing.
• No Available Members
If queuing has been enabled, calls will be queued. If queuing has not been enabled, calls will
go to the overflow group if set, even if the overflow time is not set or is set to 0. If queuing
is not enabled and no overflow is set, calls will go to voicemail. If voicemail is not available,
external calls go to the incoming call routes fallback destination while internal calls receive
busy indication.
Hunt Group Member Availability Settings
Manager Forwarding and do not disturb controls for a user are found on the User | Forwarding and
User | DND tabs.
Enabling and disabling a users hunt group membership is done by ticking or unticking the
user entry in the hunt group's extensions list on the Hunt Group | Hunt Group tab.
Controls The following short code features/button programming actions can be used:

SoftConsole A SoftConsole user can view and edit a user's settings. Through the directory, select the
required user. Their current status including DND, Logged In and hunt group membership
states are shown and can be changed. Forwarding settings can be accessed by then
selecting Forwarding.

Feature/Action Short Code Default Button

Hunt Group Enable HGEna - Toggles.
Hunt Group Disable HGDis
Forward Hunt Group On -*50 FwDH+ - Toggles
Forward Hunt Group Off -*51 FwDH-
Busy on Held BusyH
Do Not Disturb On -*08 DNDOn - Toggles
Do Not Disturb Off -*09 DNDOf
Extn Login -*35*N# Login
Extn Logout -*36 Logof

Related links
Group Operation on page 765

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 Example Hunt Group

Example Hunt Group

The follow are simple examples of how a department might use the facilities of a hunt group.
1. Basic Hunt Group
The Sales department want all sales related calls to be presented first to Jane, then Peter and
finally Anne.
Actions 1. Create a hunt group named Sales and assign it an extension number.
2. Set the Hunt Type to Sequential.
3. Add Jane, Peter and Ann to the User L ist in that order.
4. Turn off queuing on the Queuing tab and voicemail on the Voicemail tab.
5. Route relevant calls to the Sales group by selecting it as the destination in the
appropriate Incoming Call Routes.
Results Any call received by the Sales hunt group is first presented to Jane if she is available.
If Jane is not available or does not answer within 15 seconds the call is presented
to Peter. If Peter is not available or does not answer within 15 seconds the call goes
Anne. Since voicemail is not on, the call will continue to be presented around the group
members in that order until it is answered or the callers hangs up.

2. Adding Voicemail Support

A voicemail server has now been added to the system. The Sales department wants to use it
to take messages from unanswered callers. When messages are left, they want Jane to receive
message waiting indication.
Actions 1. Open the Sales hunt group settings and select Voicemail On on the Voicemail tab.
2. Select the User settings for Jane. On the Source Numbers tab, add the entry
HSales.
Results Once a call to the Sales group has been presented to all the available members, if it is
still unanswered then the call will be redirected to the group's voicemail mailbox to leave
a message. When a message has been left, the message waiting indication lamp on
Jane's phone is lit.

3. Using the Queuing Facility

The Sales department now wants calls queued when no one is available to answer. However if the
number of queued calls exceeds 3 they then want any further callers directed to voicemail.
Actions 1. Open the Sales hunt group settings and select Queuing On on the Queuing tab.
2. Set the Queue Limit to 3.
Results When the Sales group are all on calls or ringing, any further calls to the group are
queued and receive queuing announcements from the voicemail server. When the
number of queued calls exceeds 3, any further calls are routed to the group's voicemail
mailbox.

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Group Operation

4. Using Out of Service Fallback

During team meetings, the Sales department want their calls redirected to another group, for this
example Support.
Actions 1. Open the Sales hunt group settings and select the Fallback tab. In the Out of
Service Fallback Group field select the Support group.
2. Create a system short code *88/Set Hunt Group Out of Service/300.
3. Create a system short code *89/Clear Hunt Group Out of Service/300.
Results Prior to team meetings, dialing *88 puts the Sales group into out of service mode. Its
calls are then redirected to the Support group. Following the meeting, dialing *89 puts
the Sales group back In Service.

5. Using a Night Service Time Profile

Outside their normal business hours, the Sales department want their group calls automatically
sent to voicemail. This can be done using a time profile and leaving the Night Service Fallback
Group setting blank.
Actions 1. Create a Time Profile called Sales Hours and in it enter the times during which the
Sales department are normally available.
2. Open the Sales hunt group settings and select the Fallback tab.
3. In the Time Profile field select Sales Hours.
Results Outside the normal business hours set in the time profile, the Sales hunt group is
automatically put into Night Service mode. Since no Night Service Fallback Group has
been set, calls are redirected to voicemail.

Related links
Group Operation on page 765

CBC/CCC Agents and Hunt Groups

The use of and reporting on hunt groups is a key feature of call center operation. For IP Office,
reporting is provided through the Compact Business Center (CBC) or Compact Contact Center
(CCC) applications.
In order for these applications to provide hunt group and hunt group user (agent) reports, the
following rules apply:
• The hunt group names must be restricted to a maximum of 12 characters.
• The hunt group and user extension numbers should be a maximum of 4 digits.
• Hunt group members should be given a Login Code and set to Force Login.
• The agent state Busy Not Available is equivalent to Do Not Disturb. The agent state Busy
Wrap Up is equivalent to hunt group disable.

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 Coverage Groups

Related links
Group Operation on page 765

Coverage Groups
For users with a Coverage Group selected, coverage group operation is applied to all external
calls that are targeted to the user.
For external calls:
In scenarios where an external call would normally have gone to voicemail, it instead continues
ringing and also starts alerting the members of the coverage group.
• The follow me settings of Coverage Group members are used, the forwarding settings are
not.
• If the user is not available, for example if they have logged off or set to do not disturb,
coverage group operation is applied immediately.
• If the user is configured for call forward on busy, coverage operation is applied to the user's
calls forwarded to the forward on busy destination.
Coverage group operation is not applied to the following types of call:
• Hunt group calls.
• Recall calls such as transfer return, hold recall, park recall, automatic callback.
The Coverage Group is set through the user's User | Telephony | Supervisor Settings or through
their associated User Rights | Telephony | Supervisor Settings. The only group settings used are:
• The list of group members. They are treated as a collective group regardless of the group's
configuration.
• If the group has Night Server Fallback Group and or Out of Service Fallback Group set,
the members of those groups are used if the coverage group is set to night service mode or
out of service mode respectively.
Related links
Group Operation on page 765

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Chapter 75: Mobile Call Control

Mobile call control is only supported on digital trunks, including SIP trunks. It allows a user receiving
a call on their twinned device to access system dial tone and then perform dialing action including
making calls and activating short codes.
After answering a twinned call, the Mobile Call Control user can dial ** (within 1 second of each
other) to place that call on hold and instead get dial tone from the system. Any dialing is now
interpreted as if the user is logged into a basic single line extension on the system using their user
settings. That also include user BLF status indication.
To use these features the user must be configured to support mobile call control.

Warning:
• This feature allows external callers to use features on your phone system and to make
calls from the phone system for which you may be charged. The only security available
to the system is to check whether the incoming caller ID matches a configured users'
Twinned Mobile Number setting. The system cannot prevent use of these features by
caller's who present a false caller ID that matching that of a user configured for access to
this feature.
Trunk Restrictions
Mobile call control is only supported on systems with trunk types that can give information on
whether the call is answered. Therefore, mobile call control is not supported on analog or T1 analog
trunks. All other trunk types are supported (ISDN PRI and BRI, SIP (RFC2388), H323).
• Routing via trunks that do not support clearing supervision (disconnect detection) should not be
used.
• DTMF detection is applied to twinned calls to a user configured for this feature. This will have
the following effects:
• DTMF dialing is muted though short chirps may be heard at the start of any DTMF dialing.
• DTMF dialed by the user will not be passed through to other connected equipment such as IVR
or Voicemail.
Mobile Call Control Features and FNE Services
Mobile call control uses a short code set to invoke an FNE service. The codes relevant to mobile call
control are summarized below.

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FNE Description
31 Mobile Call Control
This code allows a user called or calling the system to invoke mobile call control and to then handle
and make calls as if they were at their system extension.
32 Mobile Direct Access
Mobile direct access FNE32 immediately redials on switch the DDI digits received with the call rather
than returning dial tone and waiting for DTMF digits as with FNE31 .
33 Mobile Callback
Mobile callback allows the user to call the system and then hang up. The system will then make a
call to the user's CLI and when answered, provide them with dial tone from the system to make calls.
35 Simplified Mobile Call Control
In addition to the Mobile Call Control feature that enables your mobile to make and handle calls as if
your are using your extension, this Simplified Mobile Call Control FNE 35 clears the dial tone when
the call recipient ends the call. The dial tone is provided on the mobile phone for fresh calls after the
current call is cleared.
36 Simplified Mobile Direct Access
In addition to the Mobile Direct Access feature, the Simplified Mobile Direct Access FNE36 clears
the dial tone when the call recipient ends the call.
37 Simplified Mobile Callback
In addition to the Mobile Callback feature that enables your mobile to get call back from the system
and lets you use the dial tone for making and handling calls, this Simplified Mobile Callback FNE
37 clears the dial tone when the call recipient ends the call. The dial tone is provided on the mobile
phone for fresh calls after the current call is cleared.

The codes relevant to mobility are summarized in the table.

FNE Number Feature
00 System Dial Tone
01 Steal Call
02 Auto Call Back
04 Forward All Calls
05 Forward Busy and No Answer Calls
06 Call Forward Disable
07 Park Call
08 Call UnPark
09 Pick Up Group
10 Directed Call Pick Up
12 Withheld CLI (To External Calls off IPO)
13 Enable CLI (To External Calls off IPO)
Table continues…

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Mobile Call Control

FNE Number Feature

14 Conference Add
15 Drop Call
16 Private Call (cannot be intruded or recorded)
17 Held Appearance Select
18 Same as FNE 00–Dial Tone Appearance (a=)
19 Enable Twinning
20 Disable Twinning
24 DND On
25 DND Off
26 Blind Transfer
27 Transfer to Voicemail

Using Mobile Call Control

In addition to using ** to access mobile call control, the user has access to the following additional
controls:
• Clearing a Call: *52 It may be necessary to clear a connected call, for example after
attempting a transfer and hearing voicemail or ringing instead. To do this dial ** for dial tone
and then *52 (this is a default system short code and can be changed if required).
• Return to Dial Tone: ## Return to dial tone after getting busy, number unobtainable or short
code confirmation tones from the system.
Enabling Outgoing Mobile Call Control
1. Configure the user for Mobile Twinning and Mobile Call Control On the User | Mobility
tab do the following:
• Enable Mobility Features for the user.
• Set the Twinned Mobile Number for the user's twinned calls destination.
1. Digits are matched from right to left.
2. The match must be at least 6 digits. If either the CLI or the Mobile Twinned Number is less
than 6 digits no match will occur.
3. Matching is done for up to 10 digits. Further digits are ignored. If either the CLI or Mobile
Twinned Number is less than 10 digits, matching stops at that shorter length.
4. If multiple matches occur the first user in the configuration is used. Manager will warn
against configuration where such a conflict may exist.
• Select Can do Mobile Call Control.
On systems with some unsupported trunk types, further changes such as Outgoing Group ID,
system shorts codes and ARS may be necessary to ensure that calls to the mobile twinned numbers
are only routed via trunks that support mobile call control.

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 Mobile Direct Access (MDA)

Incoming Mobile Call Control

The system can be configured to allow Mobile Call Control users to use this function when making
an incoming call to the system. This requires the user to make the incoming call from the same CLI
as their Mobile Twinning Number (even if they do not actually use Mobile Twinning).
The call will be rejected:
• If the caller ID is blank or withheld.
• If the caller ID does not match a Twinned Mobile Number of a user with Can do Mobile Call
Control enabled.
• If the call is received on a trunk type that does not support Mobile Call Control.
Enabling Incoming Mobile Call Control
On the User | Mobility tab do the following:
1. Enable Mobility Features for the user.
2. Set the Twinned Mobile Number to match the CLI of the device from which the user will be
making calls.
3. Select Can do Mobile Call Control.

Add a FNE Short Code In the system short codes section of the configuration add a short
code similar to the following. Key points are the use of the FNE Service feature and the Telephone
Number value 31.
• Short Code: *89
• Feature: FNE Service
• Telephone Number: 31

Add an Incoming Call Route for the user Create an incoming call route that matches the
user's CLI and with the FNE short code created above as its destination.
On systems with some unsupported trunk types, further changes such as Incoming Group ID
changes may be necessary to ensure that only calls received on trunks that support Mobile Call
Control are routed to this short code.
Related links
Mobile Direct Access (MDA) on page 779
Mobile Callback on page 781

Mobile Direct Access (MDA)

For a Mobile Call Control or one-X Mobile client user, FNE32 immediately redials on switch the
DDI digits received with the call rather than returning dial tone and waiting for DTMF digits as with
FNE31. This is called Mobile Direct Access (MDA).
MDA requires the user's external telephony provider to provide a direct trunk with DDI to the
system (ie. an ISDN or SIP trunk). By assigning a specific incoming line group ID to the trunk, an

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Mobile Call Control

incoming call route can be created for the same line group ID with blanks incoming number and
incoming CLI fields. The destination is a short code set to FNE32.
User validation is performed using the CLI in the same way as for normal Mobile Call Control. In
addition the call will be rejected no DDI digits are provided. Once connected the user can use the
other Mobile Call Control features such as **.

Related links
Mobile Call Control on page 776

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 Mobile Callback

Mobile Callback
Mobile callback allows the user to call the system and then hang up. The system will then make
a call to the user's CLI and when answered, provide them with dial tone from the system to make
calls.
Mobile callback is subject to all the normal trunk type and user licensing restrictions of mobile call
control. In addition the user must have the Mobile Callback (User | Mobility)setting enabled in
the system configuration.
When the user makes a call using a DDI that is routed to an FNE33 short code, the system will
not connect (answer) the call but will provide ringing while it waits for the user to hang up (after 30
seconds the system will disconnect the call).
• The system will reject the call if the CLI does not match a user configured for Mobile Callback
or does not meet any of the other requirements for mobile call control.
• The system will reject calls using FNE33 if the user already has a mobile twinning or mobile
call control call connected or in the process of being connected. This includes a mobile
callback call in the process of being made from the system to the user.
If the CLI matches a user configured for mobile callback and they hang up within the 30 seconds,
the system will within 5 seconds initiate a callback to that user's CLI.
• If the call is answered after the user's Mobile Answer Guard time and within the user's No
Answer Time, the user will hear dial tone from the system and can begin dialling as if at their
system extension.
• If the call is not answered within the conditions above it is cleared and is not reattempted.
Related links
Mobile Call Control on page 776

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Chapter 76: Transferring calls

The IP Office system supports a range of methods for transferring calls.

Related links
Transferring call notes on page 782
Transferring call notes on page 783
Off-Switch Transfer Restrictions on page 784
Context Sensitive Transfer on page 785
Dial Tone Transfer on page 786
Handsfree Announced Transfers on page 788
One Touch Transferring on page 790
Centrex Transfer on page 791

Transferring call notes

The following are some of the methods usable to transfer calls.
Note Description
Supervised This is a transfer where the user waits for the transfer destination to answer and
Transfer talks to that party before completing the transfer, this is referred to as a consultation
call. They then either complete the transfer or drop the call and return to the held for
transfer call. The call details, display, ringing and forwarding applied are appropriate
to the type of call (internal or external) being transferred.
Unsupervised This is a transfer that is completed whilst the destination is still ringing. This is also
Transfer called a 'blind transfer'.
Automatic The system allows users to automatically transfer calls using forwarding options. For
Transfer - full details, see DND, Follow Me and Forwarding on page 744.
Forwarding
Transferring to When transferring a call to another extension that has forwarding enabled, the type
a Forwarded of call being transferred is used. For example, if transferring an internal call, if the
Extension transfer target has forwarding of internal calls enabled, then the forward is used.
Table continues…

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 Transferring call notes

Note Description
Transferring Calls User's can transfer calls to their own extension number. This is useful for users with
to Yourself multiple devices registered to the same extension number or users with twinned
devices. It allows the user to transfer a call answered on one device and then
answer it on another one of their devices.
Reclaim If a transferred call is still ringing unanswered, it may be possible to reclaim the call.
The default shortcode for this is *46.
Transfer Return Sets the delay after which any call transferred by the user, which remains
Time unanswered, should return to the user. A return call will continue ringing and does
not follow any forwards or go to voicemail.
• Transfer return only occurs if the user has an available call appearance button.
• Transfer return is not applied if the transfer is to a hunt group that has queuing
enabled.

Related links
Transferring calls on page 782

Transferring call notes

The following are the basic methods for transferring calls.
Analog and single line phones
Action Steps
Unsupervised 1. Press R. Note that broken dial tone is heard while a call is on hold.
Transfer
2. Dial the transfer destination number.
3. Hang-up.
Supervised 1. Press R.
Transfer
2. Dial the transfer destination number.
3. If the destination answers and accepts the call, hang-up.
4. If the called party does not answer or does not want to accept the call, press
Ragain.
5. To return to the original caller press R.
Reclaim *46

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Transferring calls

Avaya multiple line phones

Action Steps
Unsupervised 1. Press Transfer.
Transfer
2. Dial the transfer destination number.
3. Press Transfer again to complete the transfer.
Supervised 1. Press Transfer.
Transfer
2. Dial the transfer destination number.
3. If the destination answers and accepts the call, press Transfer again to
complete the transfer.
4. If the called party does not answer or does not want to accept the call, press
Drop.
5. To return to the original caller press it’s call appearance button.
Reclaim *46

Related links
Transferring calls on page 782

Off-Switch Transfer Restrictions

Users cannot transfer calls to a destination that they cannot normally dial. This applies to manual
transfers and also to automatic transfers (forwarding). In addition to call barring applied through
short codes, the following system settings may restrict a users ability to transfer calls.
User Specific Controls
Setting Description
Outgoing Call Bar Default = Off (User | Telephony | Supervisor Settings)
When enabled, this setting stops a user from making any external calls. It therefore
stops them making any external transfers or forwards.
Inhibit Off-Switch Default = Off (User | Telephony | Supervisor Settings).
Forward/Transfer
When enabled, this setting stops the specific user from transferring or forwarding
calls externally. This does not stop another user transferring the restricted users
calls off-switch on their behalf.
• User attempts to set an external forward destination using a short code hear error
tone.
• User attempts to set an external forward destination using a programmable button
on their phone do not allow the number to be saved.

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 Context Sensitive Transfer

Line Specific Control

Setting Description
Analog Trunk to Default = Off (Line | Analog Line | Analog Options)
Trunk Connection
When not enabled, users cannot transfer or forward calls on one analog trunk back
off-switch using another analog trunk.

•
System Wide Controls
Setting Description
Inhibit Off-Switch Default = On(System | Telephony)
Forward/Transfer
When enabled, this setting stops any user from transferring or forwarding calls
externally.
• User attempts to set an external forward destination using a short code hear error
tone.
• User attempts to set an external forward destination using a programmable button
on their phone do not allow the number to be saved.
Restrict Network Default = Off (System | Telephony).
Interconnect
When this option is enabled, each trunk is provided with a Network Type option that
can be configured as either Public or Private. The system will not allow calls on
a Public trunk to be connected to a Private trunk and vice versa, returning busy
indication instead.

Conference Control
Users can use conference controls to effectively transfer calls. This includes transferring an
external call to another external number. The use of conferencing to effect off-switch transfers
can be restricted using the Inhibit External Only Impromptu Conference setting (System |
Telephony).
Related links
Transferring calls on page 782

Context Sensitive Transfer

Calls and Button Status Indication The status indication for a call on hold pending transfer has
changed to differentiate such calls from standard held calls:
• On phones with both dual lamp buttons, both the green and red lamps fast flash (flutter)
when the button represents a call on hold pending transfer.
• On phones with single lamp buttons or status icons, Xfer: is now shown in front of the caller
ID information rather than the button name. For example Xfer:Extn299 is shown rather than
a = Extn299.

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Transferring calls

• The call status information shown when the button of a call on hold pending transfer is the
currently highlight line is now prefixed with On-Hold-Xfer rather than On-Hold.
Switching Between Calls Switching from a connected call to an existing call on hold pending
transfer puts the connected call on hold pending transfer. The following table is an example of the
resulting operation .
Call or answer A Connected to A
Press Transfer A on hold pending transfer
Call or answer B A on hold pending transfer. Connected to B.
Reconnect to A Connected to A. B on hold pending transfer
Press Transferor Complete*. A transferred to B.

Requirement for a Free Call Appearance Before Starting a Transfer When the user already
has a call or calls on hold, they can now put their current call on hold pending transfer even
if there are no free call appearances available. Previously an available call appearance was
required in order to then make a consultation call to the potential transfer destination.
Conferencing Calls For these phone there have also been changes to which calls are
conferenced in different scenarios including when there is a call on hold pending transfer. See
Context Sensitive Conferencing.
Related links
Transferring calls on page 782

Dial Tone Transfer

Release 5.0+: A user who is not able to make external calls to any or some external numbers, can
be transferred to dial tone by a user who is able to make external calls.
• The restricted user wanting to make the external call, dials the unrestricted user and requests
dial tone.
• The unrestricted user initiates a transfer and dials the prefix for an ARS form configured to
provide secondary dial tone.
The prefix is a short code set up to access the required ARS form. While this can be a system
short code, using a user or user rights short code will allow control over who can provide dial tone
transfer for restricted users.
• When they hear the secondary dial tone, the unrestricted user completes the transfer.
• The restricted user now hears the secondary dial tone and is now able to make an external
call.
• The restricted user is now able to make calls as permitted by the short codes in the ARS
form.

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 Dial Tone Transfer

• The restricted user is not able to transfer the dial tone to another user.
The ARS form being used can still contain short codes that restrict the dialing that can be
attempted after the restricted user hears secondary dial tone. Other ARS features can also be
used such as alternate routing or time profiles to provide out of hours routing. The ARS form
timers are run from when the unrestricted caller dials the ARS form. They are not reset when the
restricted user is transferred to the ARS form.
Multiple prefixes and ARS forms can be used if required to create more complex scenarios. For
example, one where the unrestricted user can transfer the restricted users to an ARS forms that
allows international calls or to an ARS form that only allows national dialing.
Example Configuration:
The example below is a simple configuration that allows the unrestricted user to use 8 as a
transfer destination that provides secondary dial tone.
Create an ARS Form for Secondary Dial Tone The ARS form needs to be created before short
codes can be added to route callers to it.
• Enter a Route Name to identify the ARS form, for example Dial Tone Trans.
• Select Secondary Dial Tone.
• Select either System Tone (this matches locale specific normal dial tone) or Network Tone
(this matches locale specific secondary dial tone). For some locales both tones are the
same.
• Enter short codes that will take any digits dialed by the restricted user and process them for
external dialing to an outgoing line group. For this example we will allow any digits dialed to
be presented to the first trunk seized in outgoing line group 0.
Code N
Telephone Number N
Feature Dial
Line Group ID 0

• Other short codes can be used to allow or bar the dialing of specific numbers or types of
numbers.
• Configure the rest of the ARS form as required. For full details on ARS form configuration see
ARS.
Create a Short Code for Dial Tone Transfer For this example we will allow the prefix 8 to be
used to access an ARS form created above.
In the user short codes of the unrestricted user, create a short code that invokes the ARS form
created above. For example:
Code 8
Telephone Number
Table continues…

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Transferring calls

Feature Dial
Line Group ID 51 Dial Tone Trans

• It is important that the short code does not pass any digits to the ARS form. Once the ARS
form receives any digits, it starts short code matching and ends secondary dial tone.
• The short code could also be setup as a system or user rights short code.
The unrestricted user is now able to provide secondary dial tone to other users by on request by
pressing Transfer, dialing 8 and then pressing Transfer again.
Account and Authorization Codes:
If the restricted user enters an account or authorization code while calling the unrestricted user
to request dial tone, that value is not carried forward with their external call once they have been
provided with secondary dial tone.
If the unrestricted user enters an account or authorization code while dialing the ARS form, that
value remains associated with the call made by the restricted user.
If the ARS form short code used to route the restricted users call requires an account or
authorization code, the value already entered is used, otherwise the restricted user is prompted to
enter a value.
Call Logging:
The restricted user's outgoing call log will include the call to the unrestricted user and the outgoing
external call they subsequently make. The outgoing external call record will include the prefix
dialed by the unrestricted user to access the ARS form.
The unrestricted users call log will include just an incoming call from the restricted user.
Within the SMDR output, the calls by the restricted user are included. The call by the unrestricted
user is not included.
Related links
Transferring calls on page 782

Handsfree Announced Transfers

This feature allows the enquiry call part of a supervised transfer to be answered handsfree. In
addition the system can be optionally configured to allow both the enquiry call and completed
transfer call to be auto-answered.
Example:
1. User 201 answers a call that they then want to transfer to user 203.
2. They press Transfer to put the call on hold pending transfer.
3. They then press a Dial Direct button and dial 203.

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 Handsfree Announced Transfers

4. The transfer enquiry call is auto answered by User 203's phone. User 201 is able to
announce the pending transfer and hear if User 203 wants to accept the call.
The auto-answer only occurs if the target user's extension is idle. If the target is already connected
to a call, the transfer enquiry will be presented as normal call.
If the transfer is accepted, User 201 can press TRANSFER again to complete the transfer
process.
The transferred call will then ring at the target. However, if required the system can be configured
to also auto-answer the completed transfer.
Configuration:
Handsfree announced transfers are supported when using one of the following features after
having pressed TRANSFER.
Button Features Short Code Features
Dial Direct Dial Direct
Automatic Intercom
Dial Intercom

User Button Usability:

Following the use of any of the buttons above, if the button has not been programmed with a
specific target, a User button can be used to indicate the target for the enquiry call. This gives the
advantage of being able to see the target user's status before attempting the transfer.
• For Automatic Intercom and Dial Intercom buttons without a pre-specified target, the User
button must be on a button module.
• For Dial Direct buttons without a pre-specified target, the User button can be on the phone
or button module. Due to this and the support for Dial Direct across a network of systems,
we recommend that a Dial Direct button is used for handsfree announced transfers.
Phone Support:
Handsfree announced transfer is supported for calls being transferred to the following phones:
Full Support Partial Support Not Supported
The following system phones The following phone can auto- Announced transfer is not
support full announced transfer answer announced transfers but supported for any phones not
operation. require the user to use the listed in the other column.
handset to respond.
1603, 1608, 1616, 2410, 2420, On unsupported phones the
5410, 5420, 4610, 4621, 4625, 2402, 4601, 4602, 5402, 5601, transfer eqnuiry consultation call
5610, 5620, 5621. 5602. will be presented as a normal call.
Analog Off-Hook Stations (See
notes below).

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Transferring calls

Notes:
• On supported phones, if the target user's phone is not idle when the enquiry call attempt is
made, the enquiry call is turned into a normal transfer attempt, eg. alerting on an available
call appearance.
• Enabling the extension specific setting Disable Speakerphone will turn all auto-answer calls,
including handsfree announced transfers to the extension, into normal calls.
• Off-Hook Station Analog Phones Analog phone extensions configured as Off-Hook Station
can auto-answer transfers when off-hook and idle.
• Headset Users The following applies to users on supported phones with a dedicated
HEADSET button. These users, when in headset mode and idle will auto-answer the
announced transfer enquiry call through the headset after hearing 3 beeps. The transfer
completion will require them to press the appropriate call appearance unless they are set to
Headset Force Feed.
• Twinning Handsfree announced transfer calls to users with twinning enabled will be turned
into normal calls.
• Multi-site network Support Dial Direct is supported to targets across a multi-site network,
therefore allowing handsfree announced transfers to remote users.
Full Handsfree Transfer Operation:
If required the system can be configured to allow the full handsfree announced transfer process,
ie. both the enquiry call and the transfer, to be auto-answered on supported phones. This is done
by entering FORCE_HANDSFREE_TRANSFER into the Source Numbers of the NoUser user and
rebooting the system
Related links
Transferring calls on page 782

One Touch Transferring

This feature allows selected users to transfer calls to each other using a reduced number of key
presses.
With this option, a call can be transferred by simply selecting the transfer destination and then
hanging up (or pressing Transfer if working handsfree).
Without this option the normal sequence is to press Transfer, dial the destination and then
hanging up (or pressing Transfer if working handsfree).
For one touch transfer the transfer destination number must be selected using a button
programmed to one of the following features:
• User
• Dial

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 Centrex Transfer

• Abbreviated Dial
• Automatic Intercom
• Dial Intercom
• Dial Direct
This feature is enabled on a per user basis by adding Enable_OTT to the Source Number
settings of the user. This feature is supported on all Avaya phones that support the programmable
button features.
Related links
Transferring calls on page 782

Centrex Transfer
Centrex Transfer is a feature provided by some line providers on external analog lines. It allows
the recipient of a calls on such a line to transfer that call to another external number. The transfer
is performed by the line provider and the line is freed. Without Centrex Transfer, transferring an
external call to another external number would occupy both a incoming and outgoing line for the
duration of the call.
The following are the supported controls and usages for Centrex Transfer:
• Centrex Transfer Button Operation The action Flash Hook can be assigned to a
programmable button. This button can be configured with or without a telephone number
for an automatic or manual transfer.
- Manual Transfer If the programmable button is setup without a target telephone number,
pressing the button returns dial tone to the user. They can then dial the required transfer
number and when they hear ringing or an answer, hang up to complete the Centrex
Transfer.
- Automatic Transfer If the programmable button is setup with a target telephone number,
pressing the button performs the Centrex Transfer to the number as a single action.
• Centrex Transfer Short Code Operation The Flash Hook short code feature can be used
with system short codes. It can be setup with or without a telephone number in the same way
as a Flash Hook programmable button above. The line group must be the group of analog
lines from the Centrex service line provider.
- Centrex Transfer Operation for Analog Extensions Most analog phones have a button
that performs the action of sending a hook flash signal. The marking of the button will vary
and for example may be any of R, H, Recall or Hold. Pressing this button sends a hook
flash to the system to hold any current call and return dial tone.
• To perform a Centrex Transfer, pressing the analog extension's hook flash button should
be followed by the dialing of a Flash Hook short code.
• For analog extension users with call waiting enabled, pressing the hook flash button
during a call will hold the current call and connect any call waiting. Therefore it is

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Transferring calls

recommend that analog extension users wanting to use Centrex Transfer should not
also have call waiting enabled.
• Auto Attendant Transfer System’s using embedded voicemail can select Centrex Transfer
as an action. For system using Voicemail Pro, the equivalent can be achieved by transferring
calls to a Flash Hook short code.
Additional Notes
• Networked Systems In networked systems, Centrex Transfer is only supported using Flash
Hook or Centrex Transfer features on the system which hosts the Centrex analog trunks.
• Addition Prefix Dialing In some cases the Centrex service provider may require a prefix
for the transfer number. If that is the case, that prefix must be inserted in the button
programming or the short code used for the Centrex Transfer.
• Application Transfers Centrex Transfer is not supported for calls being held and transferred
through applications such as SoftConsole.
• Conference Calls Centrex Transfer is not supported with conference calls.
Related links
Transferring calls on page 782

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Chapter 77: Simultaneous mode

IP Office systems support 'simultaneous' mode operation. In that mode, users can be associated
with multiple telephony devices at the same time. They can answer and make calls on any of those
devices.
Related links
Simultaneous Mode Devices on page 793
Simultaneous Mode Notes on page 793
Moving Calls Between Simultaneous Devices on page 794

Simultaneous Mode Devices

An IP Office user can be logged in simultaneously on one of each of the following types of
telephone devices:
Telephony Client Notes
A physical desk phone A physical phone, including a SIP, H.323 or DECT extension. This also includes
clients running on a Vantage phone.
A desktop (PC) VoIP • Avaya Workplace Client for Windows
client:
• Avaya Workplace Client for macOS
A mobile VoIP client: • Avaya Workplace Client for Android
• Avaya Workplace Client for iOS
A WebRTC client: • Spaces Calling using the Chrome extension.

Related links
Simultaneous mode on page 793

Simultaneous Mode Notes

The following notes relate to the operation of simultaneous telephony:
• Incoming calls to the user alert on all their devices and they can choose which device they
want to use to answer.

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Simultaneous mode

• Whilst the user has a call in progress on one of the devices, any additional incoming call is
presented only to that device.
• It is recommended not to mix simultaneous mode operation with features such as such as
mobile twinning, telecommuting and mobile call controls that can lead to multiple duplicate
calls. For example, a mobile client's external PSTN numbers as a active mobile twinning
destination will cause duplicate alerts for the same call.
• Users can have their desk phone and their softphone applications registered to different
servers in an IP Office network.
• Use of simultaneous mode is not supported when also using a non-telephony CTI client to
control call handling. In that scenario it is not always possible to predict which telephony
client will be used when making/answering a call from the CTI client which can lead to
confusion.
Related links
Simultaneous mode on page 793

Moving Calls Between Simultaneous Devices

The IP Office system supports a number of features to enable users to move calls between their
simultaneous devices.
Action Description
Transfer Users can transfer calls to their own extension number. That causes the call to alert
on their other simultaneous devices.
Steal For IP Office R11.1.2.4 and higher, a Call Steal shortcode set to with the user's
extension number will retrieve a current call from their other simultaneous device.
Workplace Clients For IP Office R11.1.3 and higher, Avaya Workplace Client users can use their client
to move and retrieve calls:
• Using move, the user can send a call from their Avaya Workplace Client to their
other simultaneous devices.
• Using retrieve, the user can move a call answered on their simultaneous device to
their Avaya Workplace Client.
These features are enabled by a SET IPO_CALL_HANDOVER_ENABLED 1 line in
the 46xxsettings.txt file.

Related links
Simultaneous mode on page 793

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Chapter 78: User Source Numbers

Source numbers are used to configure features which do not have specific controls within the IP
Office Manager or IP Office Web Manager interfaces.
Sources numbers are divided into two types:
• User source numbers are used to apply settings to individual users.
• NoUser source numbers are used to apply settings to the IP Office system or to all users on
the system.
Note that the lists shown on the following pages are not exhaustive.
• Some source numbers are made obsolete when replaced by proper configuration controls in a
later release of IP Office software. At that stage, the source number is no longer supported.
• This document covers the source numbers that are publicly supported. Other source numbers
issued for particular customer sites to resolve specific issues at those sites are not included
and are not supported on other IP Office systems.
Related links
Individual User Source Numbers on page 795
NoUser Source Numbers on page 797

Individual User Source Numbers

User Source Numbers
The following source numbers affect the particular user to which they are applied. They are
mergeable unless stated otherwise.
• AT<string>
Strings beginning with AT are used with a user called DTEDefault to configure the default
settings of the control unit's DTE port.
• BST_MESSAGE_FOR_YOU
Replace the date and time shown on BST phones when idle with Message for you or
Messages for you when the user has new voicemail messages. This source number can
be used as a NoUser source number to enable the feature for all BST phone users.
• BST_NO_MESSAGE_FOR_YOU

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User Source Numbers

If the source number BST_MESSAGE_FOR_YOU has been used as a NoUser source

number to enable the feature for all BST phone users, this individual user source number can
be used to disable the feature for selected users.
• C<Conference ID>
Provides the user with message waiting indication and access to the conference mailbox of a
system meet-me conference. Access is through Visual Voice and the User Portal application.
• Enable_OTT
Enable one touch transfer operation for the user. See One Touch Transferring on page 790.
This source number can be used as a NoUser source number to enable the feature for all
users.
• H<Group Name>
Allows the user to receive message waiting indication of new group messages. The group is
added to the user’s Visual Voice menu. On suitable display extensions, the hunt group name
and number of new messages is displayed. Refer to the appropriate telephone user guide.
- If the user is not a member of the group, a voicemail code must be set for the group's
mailbox (Group | Voicemail | Voicemail Code).
• P<Telephone Number>
This record sets the destination for callback (outbound alert) calls from voicemail. Enter
P followed by the telephone number including any necessary external dialing prefix, for
example P917325559876. This facility is only available when using Voicemail Pro on which
a default or user specific Callback start point has been added. Refer to the Administering IP
Office Voicemail Pro manual. This feature is separate from voicemail ringback andVoicemail
Pro outcalling.
• R<Caller ICLID>
To allow Dial In/RAS call access only from a specified number prefix the number with a R. For
example R7325551234.
• U<User Name or Extension Number>
Allows the user to receive message waiting indication of new messages. The specified user
is added to the user’s Visual Voice menu. On suitable display extensions, the user name and
number of new messages is displayed. Refer to the appropriate telephone user guide.
- If the user is not a trusted source for the mailbox, they will need to enter its Voicemail
Code to access the mailbox.
• V<Caller ICLID>
Strings prefixed with a V indicate numbers from which access to the users mailbox is allowed
without requiring entry of the mailbox's voicemail code. This is referred to as "trusted source".
- For Voicemail Pro running in Intuity mode, trusted source is used for calls from
programmable buttons set to Voicemail Collect and Visual Voice. Other controls are
prompted for the mailbox number and then password.
Related links
User Source Numbers on page 795

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 NoUser Source Numbers

NoUser Source Numbers

The following source numbers affect all users on the IP Office system. They are entered through
the Source Numbers tab of the NoUser user. These source numbers are informally referred to as
NUSNs.
Changes to these source numbers require a system reboot to become effective.
• ATM4U_PCS7_RINGDETECT
For some cellular or mobile interfaces connected to a IP500 ATM4U card, the card may not
detect the ring signal. For PCS4 and higher card, this NoUser source number can be used
activate alternate ring detection.
• ALLOW_5410_UPGRADES
This option must be present for 5410 phones to update their firmware.
• B_DISABLE_SIP_IPADDR
Disables the blacklisting of SIP device registration based on the device IP address. Refer to
the Avaya IP Office™ Platform Security Guidelines manual.
• BST_MESSAGE_FOR_YOU
Replace the date and time shown on BST phones when idle with Message for you or
Messages for you when the user has new voicemail messages. This source number can
also be set as a source number for individual users.
• CIPHERS_LEVEL_H323=<N>
Sets the minimum cipher strength the IP Office accepts on TLS connections for H.323
phones and trunks. Not used for clients where ciphers are enabled and chosen based on
those offered by the TLS server.
- Supported for IP Office R11.1.2.x releases. For IP Office R11.3.1 and higher, this NUSN is
replaced by the System > Certificates > H.323 Security Level security setting.
- Note: The default level 1 (medium strength) is used if no source number is specified.
The value <N> is set as follows:
- Low (0) - Accept low, medium, and high-strength ciphers. Low and medium on IP500 V2
systems.
- Medium (1) - Accept medium and high-strength ciphers. Medium on IP500 V2 systems.
- High (2) - Accept high-strength ciphers. Not supported for IP500 V2 systems.
• For a list of ciphers, see https://documentation.avaya.com/bundle/IPOfficeSecurity/
page/Supported_Ciphers.html.
• High-strength ciphers are GCM ciphers. These are not supported by any model of IP500
V2 system.
• CIPHERS_LEVELS_SIP=<N>

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User Source Numbers

Sets the minimum cipher strength the IP Office accepts on TLS connections for SIP phones
and trunks. Not used for clients where ciphers are enabled and chosen based on those
offered by the TLS server.
- Supported for IP Office R11.1.2.x releases. For IP Office R11.3.1 and higher, this NUSN is
replaced by the System > Certificates > SIP Security Level security setting.
- Use the same values as CIPHERS_LEVELS_H323 but sets the cipher level the IP Office
accepts for SIP TLS connections.
• DECT_REVERSE_RING
By default, when this parameter is not set, calls on DECT phones associated with a CTI
application will ring as a priority call. When this parameter is set, DECT phones ring as a
normal, external or internal call.
• DISTINCT_HOLD_RINGBACK
Used to display a specific message about the call type for calls returning after timing out from
being parked or held. If set, such calls display Return Call - Held or Return Call – Parked
rather than connected party name or line name.
• ENABLE_J100_FQDN
Use FQDN rather than IP addresses in the server address values provided to J100 Series
phones. This requires that the FQDN values are correctly routable by the customer DNS
servers and that the phones use the DNS server address (either obtained through DHCP or
set manually).
• ENABLE_J100_AUTO_UPDATE_POLICY
Add settings for J100 Series phone auto-upgrade support to the system's auto-generated
46xxsettings.txt file. Refer to the IP Office SIP Telephone Installation Notes manual.
• Enable_OTT
Enable one touch transfer for all users. See One Touch Transferring on page 790. This
source number can also be set as a source number for individual users.
• EQNX_CONTACT_MATCHING_MIN_DIGITS=<N>
By default the Avaya Workplace Client requires at least 10 digits for contact matching (8 for
Bahrain). This NoUser source number can be used to define the minimum digits for contact
matching for countries where national dial plan phone numbers are less than 10 digits.
• FORCE_HANDSFREE_TRANSFER
If set, when using the handsfree announced transfer process (see Handsfree Announced
Transfers on page 788), both the transfer enquiry and transfer completion calls are auto-
answered. Without this setting only the transfer enquiry call is auto-answered.
• HIDE_CALL_STATE
Used to hide the call status information, for example Dial and Conn, shown on older DS
phones such as 2400, 4400 and 5400 Series. Used in conjunction with the LONGER_NAMES
source number.
• HOLD_MUSIC_TIMEOUT=<seconds>

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 NoUser Source Numbers

By default, line alternate music sources remain connected for 30 seconds after they stop
being used. You can use this source number to change the disconnect timeout. The
supported range is 1 to 600 seconds.
• LONGER_NAMES
Used to increase the length of names sent for display on older DS phones such as 2400,
4400 and 5400 Series.
• MEDIA_NAT_DM_INTERNAL=N
Used in conjunction with the setting System | VoIP | Allow Direct Media Within NAT
Location. When Allow Direct Media Within NAT Location is enabled, the default behavior
is to attempt direct media between all types of devices (H323 and SIP remote workers and
IP Office Lines behind a NAT). For routers using H323 ALG or SIP ALG, it can be desirable
to only attempt direct media between certain device types. In this case, set this NoUser user
source number where N is the sum of the following values:
- 1 = Include H323 phones.
- 2 = Include SIP phones.
- 4 = Include IP Office lines.
For example, if the router has SIP ALG that cannot be disabled, to disable attempting NAT
direct media for SIP devices, set MEDIA_NAT_DM_INTERNAL=5 to include only H323 phones
and IP Office Lines.
• NI2_CALLED.../NI2_CALLING...
The following NoUser source numbers are applied to calls on ETSI PRI trunks:
- NI2_CALLED_PARTY_PLAN=X
Forces the NI2 Called Party Numbering plan for ETSI PRI trunks, where X equals
UNKNOWN or ISDN.
- NI2_CALLED_PARTY_TYPE=X
Forces the NI2 Called Party Numbering type for ETSI PRI trunks, where X equals
UNKNOWN, INT, NATIONAL or SUBSCRIBER.
- NI2_CALLING_PARTY_PLAN=X
Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks, where X equals
UNKNOWN or ISDN.
- NI2_CALLING_PARTY_TYPE=X
Forces the NI2 Calling Party Numbering type for ETSI PRI trunks, where X equals
UNKNOWN, INT, NATIONAL or SUBSCRIBER.
• NO_DIALLED_REF_EXTERNAL
On outgoing external calls made using short codes, the short code dialed is displayed on
the user's phone and any directory matching is based on that number. This source number
changes the behavior to display the telephone number output by the short codes and base
directory matching on that number.
• onex_...

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User Source Numbers

The following NoUser source numbers are used to alter the IP addresses used for Avaya
one-X® Portal for IP Office access.
- onex_l1=<IP Address>
Sets the IP address of the one-X server that can be accessed by clients registered on the
LAN1 interface.
- onex_l2=<IP Address>
Sets the IP address of the one-X server that can be accessed by clients registered on the
LAN2 interface.
- onex_port_l1=<IP Address>
Sets the port of the one-X server that can be accessed by clients registered on the LAN1
interface.
- onex_port_l2=<IP Address>
Sets the port of the one-X server that can be accessed by clients registered on the LAN2
interface.
- onex_port_r1=<IP Address>
Sets the port of the one-X server that can be accessed by remote clients registered on the
LAN1 interface.
- onex_port_r2=<IP Address>
Sets the port of the one-X server that can be accessed by remote clients registered on the
LAN2 interface.
- onex_r1=<IP Address>
Sets the IP address of the one-X server that can be accessed by remote clients registered
on the LAN1 interface.
- onex_r2=<IP Address>
Sets the IP address of the one-X server that can be accessed by remote clients registered
on the LAN2 interface.
• PHONE_LANGUAGES
Cause an IP Office system to output a set of language files that can then be used to
customize the text used on some phones. Refer to the Avaya IP Office Locale Settings
manual.
• PRESERVED_CONN_DURATION=<Minutes (1 to 120)>
When System | Telephony | Telephony | Media Connection Preservation is enabled,
active calls are preserved for up to 120 minutes before being disconnected.. This NoUser
source number can be used to adjust the duration in the range 1 to 120 minutes.
• PRESERVED_NO_MEDIA_DURATION=<Minutes (1 to 120)>
When System | Telephony | Telephony | Media Connection Preservation is enabled, calls
on which no RTP, RTCP or speech is detected are disconnected after 10 minutes. This
NoUser source number can be used to adjust the duration in the range 1 to 120 minutes.
• PUBLIC_HTTP=<File server address>

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 NoUser Source Numbers

If the IP Office is using the HTTP Redirection settings, this source number can be used to set
a separate redirection address to be given to remote phones.
• REPEATING_BEEP_ON_LISTEN
By default, if you set Beep on Listen, when a user invokes Call Listen they hear an entry
tone (3 beeps) only at the start of the call. When this parameter is set, they also hear a beep
every 10 seconds.
• RTCP_COLLECTOR_IP=<IP Address>
When using a Prognosis server for call quality monitoring, set the IP address of the IP Office
system as configured in the Prognosis server.
• RW_SBC_...
Set the IP addresses that remote SIP extensions should use to connect to the IP Office via
an ASBCE. For R11.1.2.4 and higher, these have been replaced with settings on the System
| LAN | Network Topology menus.
• SET_46xx_PROCPSWD=<NNNNN>
Set the new password indicated to phones through the auto-generated 46xxsettings.txt
file.
• SET_96xx_SIG=<X>
When set, inserts the line SET SIG X into the auto-generated 46xxsettings.txt settings
files.
• SET_ADMINNPSWD=<NNNNN>
Set the new admin password indicated to K100 Series phones through the auto-generated
46xxsettings.txt file.
• SET_B199_FW_VER=<NNNN>
If set, overrides the default B199 firmware version the IP Office system inserts into its
auto-generated avayab199_fw_version.xml file. with firmware-NNNN-release.kt.
Supported for IP Office R11.1.2.4 and higher.
• SET_CDNL
This source number can be used to add cellular direct dialing numbers to the auto-generated
46xxsettings file. For Avaya Workplace Client clients on mobile iOS and Android devices,
this specifies numbers that should be dialed using the device's native dialer rather than using
by the client application. For details, refer to the IP Office Avaya Workplace Client Installation
Notes manual.
• SET_HEADSYS_1
If set, alters the operation of the headset button on 9600 Series phones via the auto-
generated 46xxsettings.txt settings file. Normally the headset goes off-hook when the
far end disconnects. When this option is set, the headset remains on-hook when the far end
disconnects.
• SIP_ENABLE_HOT_DESK
By default, the use of hot-desking on J129 and H175 phones is blocked. This source
numbers overrides that behavior.

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User Source Numbers

• SIP_EXTN_CALL_Q_TIMEOUT=<Minutes>
Sets the unanswered call duration after which unanswered SIP calls are automatically
disconnected. If not set, the normal default is 5 minutes. This NoUser source number can be
used to adjust the duration in the range 0 (unlimited) to 255 minutes.
• SIP_OPTIONS_PERIOD=<Minutes>
On SIP trunks, the system periodically sends OPTIONS messages to determine if the
SIP connection is active. The rate at which the messages are sent is determined by the
combination of the Binding Refresh Time (seconds) set on the Network Topology tab and
the SIP_OPTIONS_PERIOD parameter (in minutes). The frequency of sent messages is
determined as follows:
Target Method
300 seconds If no SIP_OPTIONS_PERIOD parameter is defined and the Binding
Refresh Time (seconds) is 0, then the default value of 300 seconds is
used.
Less than 300 Do not define a SIP_OPTIONS_PERIOD parameter and set the Binding
seconds Refresh Time (seconds) to a value less than 300 seconds.
More than 300 Set both the SIP_OPTIONS_PERIOD and Binding Refresh Time
seconds (seconds) to a value greater than 300 seconds.
The OPTIONS message period used is the smaller of the Binding Refresh
Time (seconds) and the SIP_OPTIONS_PERIOD.

• SET_STIMULUS_SBC_REG_INTERVAL=<seconds>
Set the registration interval used for remote J100 Series phones. Reducing this is necessary
if the SBC fails to send TCP RST end-to-end. The recommend value is 180 seconds. If not
specified, the default is 1 hour (3600 seconds). Range 180 to 3600 seconds.
• SUPPRESS_ALARM=1
When set, the NoCallerID alarm is not shown in system alarms, SysMonitor and System
Status Application .
• TUI:J139_REDUCED_FEATURE_SET
For R11.1.2.4 and higher, reinstate the pre-R11.1.2.4 feature restrictions applied to J139
phones.
• TUI:NAME_SEARCH_MODE=<n>
The default directory search matching used on feature phones is to simultaneously show
matches against all parts of names. This source number can be used to change the name
matching behavior.
- 1 = Match starting from start of name.
- 2 - Match starting from last word in name.
- 3 = Match simultaneously from both 1 & 2.
- 4 = Match from the penultimate word in name.
- 7 = Match simultaneously from first, last and penultimate words in name.

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 NoUser Source Numbers

• TUI:NO_TOVM_SK_WHEN_VMOFF
On feature phones, suppress the display of the To VM softkey when the user's VoiceMail
setting is off.
• VM_TRUNCATE_TIME=<Seconds: 0 to 7>
Analog trunks can use busy tone detection to end calls. On calls that go to voicemail, to be
recorded or to leave a message, when busy tone detection occurs, the IP Office indicates to
the voicemail server how much to remove from the end of the recording in order to remove
the busy tone segment. By default, the amount varies to match the system locale (refer to the
Avaya IP Office Locale Settings manual).
For some systems, it may be necessary to override the default if the end of analog call
recordings is either being clipped or includes busy tone. This NoUser source number can be
used to adjust the amount removed in the range 0 to 7 seconds.
• VMAIL_WAIT_DURATION=<Milliseconds>
Sets the number of milliseconds to system waits before passing call audio to Voicemail. On
some systems, a delay may be required to allow completion of codec negotiation.
• VMPRO_OOB_DTMF_OFF
Disable the sending of out-of-band digits to the Voicemail Pro voicemail server. This may be
necessary on some systems if digit presses are being recorded on calls.
• WEBRTC_...
These source numbers are used for WebRTC support when the User Portal user connects
to the remotely using either STUN and/or TURN. For R11.1.2.4 and higher, these have been
replaced with settings on the System | LAN | Network Topology menus.
• xmpp_port...
• These NoUser source numbers can be used Avaya one-X® Portal for IP Office to alter the
ports used for XMPP connections.
- xmpp_port_l1=<Port>
Set the port of the XMPP server used by clients registered on the LAN1 interface.
- xmpp_port_l2=<Port>
Set the port of the XMPP server used by clients registered on the LAN2 interface.
- xmpp_port_r1=<Port>
Set the port of the XMPP server used by remote clients registered on the LAN1 interface.
- xmpp_port_r2=<Port>
Set the port of the XMPP server used by remote clients registered on the LAN2 interface.
Related links
User Source Numbers on page 795

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Chapter 79: Configuring IP Office Server
Edition System Settings

This section covers general configuration tools for IP Office Server Edition systems.
Related links
Synchronizing Server Edition passwords in Web Manager on page 804
Creating a common administration account on page 805
Voicemail Pro Administration on page 806
Server Edition Resilience on page 806
Synchronizing the Configurations on page 806
Starting Web Control on page 807

Synchronizing Server Edition passwords in Web Manager

From IP Office Web Manager, you can open the IP Office Server Edition solution in IP Office
Manager using the Applications > IP Office Manager command. To do that, all the IP Office
systems in the solution must have a service user with common credentials.
If any IP Office system does not have a service user with the required common credentials,
opening IP Office Manager fails.
• This process synchronizes the security service users and their service user passwords on all
systems.
• This process affects service users and their passwords. It does not affect any other security
settings, including the settings of rights groups.
Before you begin
You must know the user ID and password of the service user that is common to all systems in the
solution.
Procedure
1. For the IP Office system with reset security settings:
a. Open IP Office Web Manager using the address https://<ip_address>:7070/
WebManagement/WebManagement.html where <ip_address> is the address of
the individual IP Office.

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 Creating a common administration account

b. Log on as Administrator.
c. Select Security > Service Users.
d. Change the settings of the service user used for common log in to match the settings
configured on the other IP Office systems in the solution. If necessary, create a new
user.
e. Log out of this Web Manager session.
2. For the whole IP Office Server Edition solution:
a. Open IP Office Web Manager using the address https://<ip_address>/
index.html where <ip_address> is the address of the primary IP Office.
b. Log in as the common service user.
c. Select Security > Service Users.
d. Click Synchronize Service User and System Password.
Related links
Configuring IP Office Server Edition System Settings on page 804

Creating a common administration account

About this task
When managing multiple IP Office systems, it is useful to create a common username and
password on all the IP Office systems. You can do that using the Server Edition Service User
Management tool. This tool creates a common service user called SCN_Admin.
• This process requires you to have a service user username and password for security
settings access to each of the IP Office systems.
Procedure
1. Select Tools > Server Edition Service User Management.
2. Select the IP Office systems for which you want to create a common configuration account.
3. Click OK.
4. IP Office Web Manager requests a username and password for security configuration
access to each IP Office system.
• To use the same values for all the IP Office systems, select Use above credentials for
all remaining, selected IPOs.
5. Click OK.
6. IP Office Web Manager lists the IP Office systems and whether each system already has
an SCN_Admin account.
7. Click Create Service User.

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Configuring IP Office Server Edition System Settings

8. Enter the common password and click OK.

9. Click Close.
Related links
Configuring IP Office Server Edition System Settings on page 804

Voicemail Pro Administration

If the Voicemail Pro client application is installed on the same PC as IP Office Manager, you can
launch it from IP Office Manager.
Procedure
1. In IP Office Manager, select the Solution Edition Solution View.
2. Select the IP Office server for which you want to administer the Voicemail Pro application.

3. Click on the Voicemail Administration link on the right.

Related links
Configuring IP Office Server Edition System Settings on page 804

Server Edition Resilience

For a network of IP Office systems, you can configure support for continued operation if one of
the IP Office systems becomes unavailable. For example, IP phones registered on one IP Office
system can temporarily re-register on another IP Office system and continue to make and receive
calls.
See the IP Office Resilience Overview manual.
Related links
Configuring IP Office Server Edition System Settings on page 804

Synchronizing the Configurations

About this task
By default, when configuring an IP Office Server Edition solution, the IP Office automatically
synchronizes changes to certain records across the configuration of every IP Office system in the
solution. IP Office does this for Incoming Call Route, Time Profile, Account Code and User
Rights records.

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 Starting Web Control

However, when you add a new IP Office to the solution, or you directly change the configuration of
an IP Office system, some records can become unsynchronized from the other IP Office systems
in the solution. If that happens, you can use the following process to resynchronize the shared
records.
Procedure
1. In the Server Edition Solution View, right-click on Solution.
2. Select Synchronize Configurations.
3. Select Yes to confirm the removal.
Related links
Configuring IP Office Server Edition System Settings on page 804

Starting Web Control

About this task
The web control menus are a set of web-based administration menus used by Linux-based IP
Office servers. The menus provide functions such as stopping and starting individual services run
by the IP Office server.
Procedure
1. Select the Server Edition Solution View.
2. Select the system for which you want to display its web control menus.

3. Click the Web Control link on the right.

4. The default web browser starts with the address of the IP Office system.
5. Log in using the same username and password as used for IP Office Manager
configuration access.
Related links
Configuring IP Office Server Edition System Settings on page 804

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Part 7: Small Community
Networking

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Chapter 80: Small Community Networking

You can link multiple IP500 V2 IP Office systems to form a multi-site network called a "Small
Community Network" (SCN). Within a SCN, the separate IP Office systems automatically learn each
other's extension numbers and user names. This allows calls between systems and support for a
range of internal call features (see Telephone Features Supported Across Server Edition and SCN
Networks on page 71.
Capacity
The following are the supported capacity limits for a Small Community Network system.
Maximum Number of Systems 32
Maximum Number of Users 1000
Maximum H.323 Line Hops Between Systems 5

Configuration Summary
To set up a Small Community Network, the following are required:
• A working IP Office Line trunk between the systems, that has been tested for correct voice
and data traffic routing. The arrangement the IP Office Line trunks must meet the requirements
detailed in Supported Small Community Network Layouts on page 810.
• Within a particular system, all SCN trunks should be on the same LAN interface.
• VCM channels are required in all systems.
• The extension, user and group numbering on each system must be unique.
• The user and group names on each system must be unique.
• We also recommend that all names and numbers (line, services, etc) on the separate systems
are kept unique. This will reduce potential maintenance confusion.
• The Outgoing Group ID on the Small Community Network lines should be changed to a
number other than the default 0.
• All systems should use the same set of telephony timers, especially the Default No Answer
Time.
• Check that all systems in the network are configured to use the same Codecs.
• Only one system should have its Voicemail Type set to Voicemail Pro/Lite. All other systems
must be set to either Centralized Voicemail or Distributed Voicemail. No other settings are
supported.

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Small Community Networking

Software Level Interoperation

A Small Community Network is supported between IP Office systems with the same major software
level or one level of difference in major software level. For example, between 9.1 and 9.0 (same
major level) and between 8.0 and 9.0 (one major level of difference).
• This option is intended mainly to allow the phased upgrading of sites within a Small Community
Network.
• It is still recommended that all systems within a network are upgraded to the same level where
possible.
• In a Small Community Network that includes differing levels of software, the network features
and capacity supported are those of the lowest level of software within the network.
Related links
Supported Small Community Network Layouts on page 810
Telephone Features Supported Across Server Edition and SCN Networks on page 71
SCN Voicemail Support on page 813
Enabling Small Community Networking on page 813
Small Community Network Remote Hotdesking on page 815
Small Community Network Fallback on page 816
SCN Short Code Programming on page 817

Supported Small Community Network Layouts

The allowed arrangement of IP Office Lines between the systems depends on the lowest software
level of any system in the network. Note that this refers to IP Office Lines configured in the
system configurations. The actual IP network configuration, including IP routes in the system
configurations, can differ as per the customer network requirements.
Star/Serial Layouts
The following are examples of star and serial layouts.

---- = IP network, | = IP Office Line.

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 Telephone Features Supported Across Server Edition and SCN Networks

Mesh Layout
A mesh layout is one where there is more than one possible IP Office Line route between any
two systems. The following are examples of mesh layouts. Mesh, star and serial layouts can be
combined.

Small Community Network Signaling

Within the Small Community Network, each IP Office uses signaling similar to RIP to update each
other systems of their presence.
• This traffic can be seen in the System Monitor application as AVRIP packets.
• This traffic is sent to port 50795 on which each system listens.
• Each system in the Small Community Network transmits an update every 30 seconds.
• Additionally, BLF updates are transmitted when applicable up to a maximum of every 0.5
seconds.
• Typically the volume is less than 1Kbps per system.
Related links
Small Community Networking on page 809

Telephone Features Supported Across Server Edition and

SCN Networks
In addition to sharing user and hunt group extension numbers, the following additional telephony
features are supported between systems in a multi-site network. Features not listed are not
supported across the multi-site network.
• Absence Text
• Advertised Hunt Groups Hunt groups set to advertised can be dialed by users on other
systems
• Anti-tromboning Calls routed across the multi-site network and back to the originating
system are turned back into internal calls on the originating system only.
• Break Out Dialing
• Call Park / Unpark Call
• Call Pick-up Extension

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Small Community Networking

• Call Tagging
• Callback When Free
• Centralized Call Log
• Centralized Personal Directory
• Conference
• Distributed Hunt Groups
• Distributed Voicemail Server Support
When using Vociemail Pro, each system can support its own Voicemail Pro server.
• Enable ARS / Disable ARS
• Extension Dialing
Each system automatically learns the user extension numbers available on other systems
and routes calls to those numbers.
• Resiliency Options
• Fax Relay
• Follow Me Here / Follow Me To
• Forwarding
• Hold
Held calls are signalled across the network.
• Internal Twining
• Intrusion Features
• Mobile Call Control
Licensed mobile call control users who remote hot desk to another system take their licensed
status with them.
• Music On Hold Source Selection
• Remote Hot Desking
• Set Hunt Group Out of Service / Clear Hunt Group Out of Service
• Transfer
Calls can be transferred to network extensions.
• User DSS/BLF
Monitoring of user status only. The ability to use additional features such as call pickup
via a USER button will differ depending on whether the monitored user is local or remote.
Indication of new voicemail messages provided by SoftConsole user speed dial icon is not
supported.
• User Profile Resilience

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 SCN Voicemail Support

When a user hot desks to another system, they retain their Profile settings and rights.
Related links
Working with the Server Edition Manager User Interface on page 65
Small Community Networking on page 809

SCN Voicemail Support

Within an SCN, a single Voicemail Pro is used to store all voicemail messages and provide access
to mailboxes. This Voicemail Pro server is licensed and hosted by a chosen central IP Office
system.
The use of additional Voicemail Pro servers is supported to provide call recording and auto
attendant functions to their local system. However, the central Voicemail Pro server is still used as
the message store for all messages. Refer to the Administering IP Office Voicemail Pro manual.
• User mailboxes.
• Call recording.
Recording of incoming call routes is only supported for destinations on the same system, not
for remote Small Community Network destinations.
• Dial by Name.
• Auto Attendants.
• Breakout
Requires that the numbers used are routable by the system hosting the voicemail server.
• Announcements
• UMS Web Services
Users for UMS Web Services (IMAP and or web voicemail) are licensed through the UMS
Web Services license on their host system. This applies even if the user remote hot desks to
another system in the Small Community Network.
Related links
Small Community Networking on page 809

Enabling Small Community Networking

The process below adds an IP Office Line to the system configuration. It is assumed that data
routing between the systems has already been configured and tested. Adding Small Community
Network connections between systems can also be done using Manager's Small Community
Network Management mode.

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Small Community Networking

Related links
Small Community Networking on page 809

Setup the VoIP Line from System A to System B

About this task
Receive the system configuration for System A. Prepare the system for addition to the Small
Community Network:
Procedure
1. Change all extensions numbers and names to values that will be unique within the multi-
site network.
• For users and extensions this can be done using the Extension Renumber tool. That
will adjust all users and extension and all items using those numbers, for example hunt
group memberships and incoming call routes.
• For hunt groups, each hunt group must be change individually.
2. Click Line to display a list of existing lines.
3. Right-click on the displayed list and select New and then IP Office Line.
4. Select the Line tab and set the following:
• In the Transport Type field, select Proprietary.
• In the Networking Level field, select SCN.
• In the Description field, enter a description of the link. For example System B Small
Community Network.
• Set the Outgoing Group ID to a unique value. For example match the automatically
assigned Line Number value.
5. Under Gateway, set the following:
• For the Gateway IP Address, enter the IP address of the remote System B.
• Use of IP Office SCN - Fallback is detailed in Small Community Network Fallback.
6. Click the VoIP Settings tab.
• Select the preferred Compression Mode. The same mode must be used by all VoIP
lines and extensions within the network.
• The other option can be configured as required but must be matched by the other IP
Office Lines in the network. For example the Silence Suppression settings on all the
network trunks must match.
7. Select System | Voicemail.
a. Only one system should have its Voicemail Type set to Voicemail Pro/Lite.
The Voicemail IP Address will be the IP address of the central voicemail server PC.

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 Small Community Network Remote Hotdesking

b. Any other system with its own Voicemail Pro server PC should have its Voicemail
Type set to Distributed Voicemail.
The Voicemail IP Address should be the IP address of the distributed voicemail
server PC. The Voicemail Destination should be set to the Outgoing Group ID used
for the Small Community Network line to the system that is set as Voicemail Pro/Lite.
c. All other systems should have their Voicemail Type set to Centralized Voicemail.
The Voicemail Destination should be set to the Outgoing Group ID used for the
Small Community Network line to the system that is set as Voicemail Pro/Lite.
8. Save the configuration and reboot System A.
Next steps
Set up the IP Office Line from B to A.

Setup the VoIP Line from System B to System A

Procedure
1. On the remote system, repeat the previous steps to create an IP Office Line to System A.
As stated above, where possible the line settings, especially the VoIP settings, must match
those used for other IP Office Lines in the network.
2. Load the configuration and reboot the remote system.
Next steps
Test by making calls between extensions on the different systems

Small Community Network Remote Hotdesking

The system supports hotdesking between systems within a network.
In the descriptions below, the system on which the user is configured is termed their 'home'
system, all other systems are 'remote' systems.
When a user logs in to a remote system:
• The user's incoming calls are rerouted to that system.
• The user's outgoing calls uses the settings of the remote system.
• The user's license privileges move with them, for example their user profile setting is
retained. The host system does not need to be licensed for the user.
• The user's own settings are transferred. However, some settings may become unusable or
may operate differently.
• User rights are not transferred to the remote system but the name of any user rights
associated with the user are transferred. If user rights with the same name exist on the
remote system, then they will be used. The same applies for user rights applied by time
profiles, if time profiles with the same name exist on the remote system .

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Small Community Networking

• Appearance buttons configured for users on the home system will no longer operate.
• Various other settings may either no longer work or may work differently depending on the
configuration of the remote system at which the user has logged in.
• The rights granted to the user by their Profile settings are retained by the user. There is no
requirement for the remote system to have the appropriate licenses for the Profile.
If the user's home system is disconnected while the user is remotely hot desked, the user will
remain remotely hot desked. They can remain in that state unless the current host system is
restarted. They retain their license privileges as if they were on their home system. Note however
that when the user's home system is reconnected, the user may be automatically logged back
onto that system.
Break Out Dialing In some scenarios a hot desking user logged in at a remote system will want
to dial a number using the system short codes of another system. This can be done using either
short codes with the Break Out feature or a programmable button set to Break Out. This feature
can be used by any user within the multi-site network but is of most use to remote hot deskers.
Related links
Small Community Networking on page 809

Small Community Network Fallback

Each system in the Small Community Network can include one IP Office Line where the SCN
Resiliency Options is set to Supports Resiliency. The system to which the IP Office Line
connects is then requested to provide fallback support for selected options for the local system.
• Note that both ends of the SCN trunk connection must be set to fallback.
• On the system requesting backup, the required SCN Resiliency Options are selected,
indicating that it is requesting backup. A single system can only request backup from one
other system.
• A system providing backup can provide backup for up to 7 other systems.
Resiliency is supported on Server Edition systems for Avaya 1600 and 9600 series H.323 phones.
IP500 V2 systems also support 4600 and 5600 series phones. Resiliency is configured on Line |
IP Office Line | Line under SCN Resiliency Options. The options supported are:
• Backs up my IP Phones
• Backs up my Hunt Groups
• Backs up my Voicemail
• Backs up my IP DECT Phones
Phone Resiliency
When Backs up my IP Phones is selected, the local system shares information about the
registered phones and users on those phones with the other system. If the local system is no
longer visible to the phones, the phones reregisters with the other system.

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 SCN Short Code Programming

Failback Recovery: If the setting System | Telephony | Telephony | Phone Failback is set
to Automatic, and the phone’s primary gatekeeper has been up for more than 10 minutes, the
system causes idle phones to perform a failback recovery to the original system.

Notes
• Fallback handover takes approximately 3 minutes. This ensure that fallback is not invoked
when it is not required, for example when the local system is simply being rebooted to
complete a non-mergeable configuration change.
• Fallback is only intended to provide basic call functionality while the cause of fallback
occurring is investigated and resolved. If users make changes to their settings while in
fallback, for example changing their DND mode, those changes will not apply after fallback.
• If the fallback system is rebooted while it is providing fallback services, the fallback services
are lost.
• Fallback features require that the IP devices local to each system are still able to route data
to the fallback system when the local system is not available. This will typically require each
system site to be using a separate data router from the system.
• When an IP Phone re-registers to a secondary IP Office on the failure of the primary control
unit, the second system will allow it to operate indefinitely as a “guest”, but only until the
system resets. Licenses will never be consumed for a guest IP phone.
• Remote hot desking users on H323 extensions are automatically logged out.
Related links
Small Community Networking on page 809

SCN Short Code Programming

In a multi-site network, the systems automatically learn each others extension numbers and route
calls appropriately. However the same does not apply to dialing of other numbers. Using short
codes it is possible to have the dialing of numbers on one system to be redirected to another
system and dialed there. The dialing is then matched against the short codes available on the
remote system.
Scenario
We want a short code on System A which will correctly route any 3000 range number to System
B. This will allow System B group numbers to be dialed from System A. To achieve the above
scenario, we will add a new system short code. By using a system short code it becomes available
to all users.
Example Short Code
In the configuration for System A.
1. Click Short codeto display a list of existing system short codes.
2. Right-click on the displayed list and select New.

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Small Community Networking

3. Enter the short code settings as follows:

• Short Code: 3XXX This will match any four-digit number beginning with 3.
• Telephone Number: . The . indicates that the short code should output the digits as dialed.
• Line Group ID: 99999 This should match the Outgoing Group ID given to the IP Office Line
connected to System B.
• Feature: Dial
Click OK.
A similar system short code can be added to System B's configuration to route 2XXX dialing to
System A.
Related links
Small Community Networking on page 809

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Chapter 81: Small Community Network
Management

Manager supports the ability to load and manage the configurations of the systems in a Small
Community Network at the same time. Manager must be enabled for Small Community Network
discovery.
When the configurations of the systems in a Small Community Network are loaded, Manager
switched to Small Community Network management mode. This differs from normal system
configuration mode in a number of ways:
• A network viewer is available. In addition to giving a graphical view of the Small Community
Network, the view can be used to add and remove links between the systems in the Small
Community Network.
• In the configuration tree, the records for users and hunt groups on all systems are grouped
together.
• Time Profiles and User Right common to all systems are grouped together.
• The configuration settings for each system in the Small Community Network can be accessed
and edited.
Related links
Enabling SCN Discovery on page 819
Creating a Common Admin Account on page 820
Loading a Small Community Network Configuration on page 821
Editing a Small Community Network Configuration on page 822
System Inventory on page 823

Enabling SCN Discovery

About this task
In order for the Select IP Office menu to groups systems in a Small Community Network and
allow loading of all the Small Community Network configurations, Manager must be enabled for
SCN discovery.

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Small Community Network Management

Procedure
1. Select File | Preferences.
2. Select the Discovery tab.
3. Select the SCN Discovery option.
4. Check that the other discovery setting are sufficient to discover all the systems in the Small
Community Network.
5. Click OK.
Related links
Small Community Network Management on page 819

Creating a Common Admin Account

About this task
When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Select Tools | SCN Service User Management.
The option is not shown if a Basic Mode system configuration is loaded. If no configuration is
loaded, and the option is not shown, select View | Advanced View.
Procedure
1. The Select IP Office menu displays the list of discovered systems.
2. Select the systems for which you want to create a common configuration account.
Click OK.
3. A user name and password for security configuration access to each system is requested.
Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials
for all remaining, selected IPOs.
4. The systems will be listed and whether they already have an SCN_Admin account is
shown.
5. To create the SCN_Admin account on each system and set the password for those
account click on Create Service User.
6. Enter the common password and click OK.

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 Loading a Small Community Network Configuration

7. The password can be changed in future using the Change Password option.
8. Click Close.
Related links
Small Community Network Management on page 819

Loading a Small Community Network Configuration

About this task
If Manager is configured with SCN Discovery enabled, the Select IP Office menu will display any
SCNs it discovers.
Procedure
1. With no configuration loaded, click on or select File | Open Configuration.
2. The Select IP Office menu is displayed.
Any systems in a Small Community Network will be grouped together.
3. To load the configuration of all the systems in the network, click the check box next to the
network name and then click OK.

If a warning icon is displayed next to the SCN check box, it indicates that not all
the systems known to be in the Small Community Network were discovered. Hovering
the cursor over the icon will display details of the missing systems. Loading the network
configuration at this time would not include the configuration of the missing system or
systems. The missing systems:
• May be disconnected
• The discovery settings for the Manager PC may be incorrect.
• The data routing between the Manager PC and the missing systems may be incorrect or
blocked.
4. Enter the name and password for configuration access to each system.
If the systems all have a common user name and password (see Common Administrator
Access below), select Use above credentials for all remaining selected IPOs. Click OK.
5. Manager will load and display the combined configurations in Small Community Network
Management mode.
Related links
Small Community Network Management on page 819

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Small Community Network Management

Editing a Small Community Network Configuration

When the configuration of a Small Community Network is loaded, Manager displays the
configuration in a different way from when the configuration of a single system is loaded. The
main differences are in how configuration records are grouped in the configuration tree.

Clicking on the Small Community Network icon displays the Network Viewer which shows the
lines between the systems in the Small Community Network.
• Small Community Network Configuration Records
Certain records from each of the systems in the Small Community Network are grouped
together in the configuration tree differently from when just a single system configuration is
loaded. There are two types, unique Small Community Network records and shared Small
Community Network records:
• Unique Records
They can be edited here and the system to which they belong is indicated in the group pane
and in the title bar of the details pane. However, to add or delete these types of record must
be done within the configuration records of the particular system that will host the entry's
configuration details.

- All user in the Small Community Network are shown under the User icon.

- All hunt groups in the Small Community Network are shown under the Hunt Group
icon.
• Shared Records
Shared records are configuration items that exist on all systems in the Small Community
Network, having the same name and settings on each system. Editing the shared record
updates the matching copy in the configuration of each system. Similarly, adding or deleting
a shared record adds or deletes from the individual system configurations. If the copy of the
shared record within an individual configuration is edited, it is no longer a shared record for
the Small Community Network though the individual records on other system will remain.
Changing the individual records back to matching will turn the records back into a shared
record.

- Shared time profiles are shown under the Time Profile icon.

- Shared user rights are shown under the User Rights icon.

• Individual System Configurations

The full configuration for each system in the Small Community Network can be accessed and
edited as required. It is possible to copy and paste configuration records between systems
using the configuration tree.

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 System Inventory

Saving Changes
When the save icon or File | Save Configuration is selected, the menu for multiple
configuration saves is displayed. It provides similar options are for a normal single configuration
save. Note that when working in Small Community NetworkManagement mode, after saving
configuration changes the Manager will always close the displayed configuration.
• Change Mode
If Manager thinks the changes made to the configuration settings are mergeable, it will select
Merge by default, otherwise it will select Reboot.
- Merge
Send the configuration settings without rebooting the system. This mode should only be
used with settings that are mergeable.
- Reboot
Send the configuration and then immediately reboot the system.
- Reboot When Free
Send the configuration and reboot the system when there are no calls in progress. This
mode can be combined with the Call Barring options.
- Timed
The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
• Reboot Time
This setting is used when the reboot mode Timed is selected. It sets the time for the system
reboot. If the time is after midnight, the system's normal daily backup is canceled.
• Call Barring
These settings can be used when the reboot mode Reboot When Free is selected. They bar
the sending or receiving of any new calls.
• Error Status
The warning will appear if the configuration being sent contains any validation errors
indicated by a icon in the error pane. The configuration can still be sent if required.
Related links
Small Community Network Management on page 819

System Inventory
When working in Small Community Network Management mode, clicking on the System icon for a
particular system displays a system inventory page for that system.

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Small Community Network Management

Related links
Small Community Network Management on page 819

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Chapter 82: Using the Network Viewer

Clicking on Small Community Network in the configuration tree displays the Network Viewer. This
shows each of the systems in the Small Community Network and the links between each of the
systems. Systems with attached Voicemail Pro servers are also indicated.

Green System with Voicemail Pro system.

Black Small Community Network line between two systems.
Red Incorrect Small Community Network line between systems (probably one-way connection).
Right-click on the line and select Repair.

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Using the Network Viewer

You can use the Network Viewer to perform a range of functions:

• Arrange the View
• Launch System Status
• Launch Voicemail Pro
• Add an IP Office Line
• Add a system
• Remove an IP Office Line
• Remove a system from the Small Community Network
• Repairing an IP Office Line
• Add a Background Image

Arranging the View

About this task
Procedure
You can click and drag items around in order to position them where required.
Alternatively if you right click on the view you can select Auto Arrange.
Note:
The position of elements in the network view are stored as part of the system configuration.
Therefore changes to the view will require the configuration to be saved.

Adding a Line Within the Small Community Network

About this task
You can use the network viewer to add a Small Community Network link between two systems in
the Small Community Network that are currently linked. This process will add new H.323 Small
Community Network line records to the configurations of each of the systems.
Note:
Adding a line between systems will require those systems to reboot when the changes are
saved.
Procedure
1. Right click on the starting system for the link.
Select Connect To and select the name of the other system in the Small Community
Network to which you want to link.

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 Repairing a Line Within the Small Community Network

2. Select the type of line, IP Office SCN or IP Office SCN-Fallback.

3. Click OK.
If Small Community Network-Fallback is selected, the actual backup function still need to
be configured.
a. The newly added line is displayed in the network viewer.
b. Click OK.

Repairing a Line Within the Small Community Network

About this task
A red line in the network viewer indicates a incorrectly configured line between two systems in the
Small Community Network. Typically this will be a line configured in one of the systems but not
matched by a line configured in the other system. The network viewer can be used to correct this
error.
Procedure
1. Note that adding a line between systems will require those systems to reboot when the
changes are saved.
2. Right click on the red line and select Repair Line.
3. The line is changed to black.
4. Click OK.

Adding a System to the Small Community Network

About this task
You can use the network viewer to add a Small Community Network line to a system not yet in
the Small Community Network. This process will add new H.323 Small Community Network line
records to the configurations of each of the systems.
Procedure
1. Note that adding a line between systems will require those systems to reboot when the
changes are saved.
2. Right click on the starting system for the link.
Select Connect To and select Discovery.
3. The Select IP Office menu will display any discoverable systems not already in the Small
Community Network.

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Using the Network Viewer

If the discovery includes systems already in another Small Community Network it will not
indicate such. If you want to add such a system in order to join the SCNs you can do
so. However after adding the system, you should immediately save the configuration and
reload the Small Community Network.
a. Select the required system and click OK.
b. Enter the name and password for configuration access to the selected system and
click OK.
c. The newly added system is displayed in the network viewer.
d. Click OK.
The configuration of the newly added system is now included in the configuration tree.
e. If the Error List is visible (View | Error Pane), check that none of the error are
Small Community Network specific errors, for example duplicate names or extension
numbers.

Removing a Small Community Network Line

About this task
You can use the network viewer to remove the Small Community Network lines between two
systems in the Small Community Network.
Procedure
1. Note that removing a link between systems will require those systems to reboot when the
changes are saved.
2. Right click on the link and select Delete Line.
3. The line is removed from the network viewer.
4. Click OK.

Removing a System
About this task
You can use the network viewer to remove a system from the Small Community Network.
Procedure
1. Note that removing a system will require previous linked systems to reboot when the
changes are saved.
2. Right click on the system and select Remove From Small Community Network.

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 Start System Status

3. Any lines to other system in the Small Community Network are removed.
4. Click OK.

Start System Status

About this task
If the System Status Application is also installed on the Manager PC, you can start it for a
particular system.
Procedure
1. Right click on the system and select System Status.
2. The application is started and the login form pre filled with the IP address of the system.

Start Voicemail Pro

About this task
If the Voicemail Pro client is also installed on the Manager PC, you can start it for the any system
with an associated Voicemail Pro server.
Procedure
Right click on the voicemail server icon and select Launch VMPro Client.

Add a Background Image

About this task
You can select an image file to be displayed in the background of the Network Viewer display. This
file is not saved as part of the configuration in any way, ie. if the image file is moved or deleted it is
not longer used by Manager.
Procedure
1. Right click on the general background area of the network viewer and select Background
Image.
2. Select Set Background Image to browse to the location of the file to be used.
3. The Visible option can be used to switch the display of the background image on or off.

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Part 8: SIP Trunks

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Chapter 83: SIP Trunk Overview

A growing number of service providers now offer PSTN access to businesses via public SIP trunk
connections, either to extend their reach beyond their typical copper based network coverage areas,
or so that multiple services (voice and internet access) can be bundled into a single network
connection. Although detailed public SIP trunk service offerings vary depending on the exact nature
of the offer from the specific service provider, SIP trunks can potentially provide several advantages
compared to traditional analog or digital trunks. These advantages include:
• cost savings resulting from reduced long distance charges, more efficient allocation of trunks,
and operational savings associated with managing a consolidated network
• simplified dialing plans and number portability
• geographic transparency for local accessibility creating a virtual presence for incoming calls
• trunk diversity and redundancy
• multi-media ready to roll out future SIP enabled applications
• fewer hardware interfaces to purchase and manage, reducing cost and complexity
• faster and easier provisioning
IP Office delivers functionality that enhances its ability to be deployed in multi-vendor SIP-based
VoIP networks. While this functionality is primarily based on the evolving SIP standards, there is
no guarantee that all vendors, interpret and implement the standards in the same way. To help the
SIP service provider, Avaya operates a comprehensive SIP Compliance Testing Program referred to
as GSSCP. Avaya's DevConnect program validates the operation of the IP Office solution with the
service provider’s SIP trunk offering.
Related links
Configuring a SIP Trunk on page 831
SIP Line Requirements on page 832

Configuring a SIP Trunk

This procedure provides the basic steps for configuring a SIP trunk between two IP Office
systems.
Before you begin
• You must know the IP address of both ends of the trunk.

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SIP Trunk Overview

• You must have valid licenses on both IP Office systems.

• On Server Edition, make sure you have a non-zero value in the SIP Trunk Sessions field
on the License | Remote Server tab. If you do not, you will see Monitor messages about
insufficient licenses.
Procedure
1. In the Manager navigation pane, right click Line and select New > SIP Line.
2. Record the Line Number value that appears on the SIP Line page for use later.
3. In the ITSP Domain Name field, enter the domain name required by the far end. If nothing
is configured in this field, then IP Office inserts the far end's ITSP Proxy Address from the
Transport tab as the ITSP domain in the SIP messaging.
4. Use the default values for the remaining fields.
5. Select the Transport tab.
6. In the ITSP Proxy Address field, enter the IP Address of the far end.
7. Select the SIP URI tab.
8. Click Add.
9. Enter values for the Incoming Group and Outgoing Group fields. You can use the Line
Number from the SIP Line tab for both values.
10. In the Manager navigation page, select Incoming Call Route.
11. On the Standard tab, in the Line Group ID field, enter the Line Number from the SIP
Line tab.
12. Select the Destinations tab.
13. In the Destination column, replace the value with a period (“.”).
14. In the Manager navigation pane, select Short Code.
15. Add a short code to dial the trunk you have just added.
16. One end of the trunk is now configured. Save the configuration to the IP Office.
17. Using Manager, open the configuration for the IP Office at the other end of the SIP trunk
and repeat the steps.
Related links
SIP Trunk Overview on page 831

SIP Line Requirements

Use of SIP requires the following:
• SIP Service Account

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 SIP Line Requirements

An account or accounts with a SIP internet service provider (ITSP). The method of operation
and the information provided will vary. The key requirement is a SIP URI, a web address of
the form [email protected]. This is the equivalent of a SIP telephone number for making
and receiving calls via SIP.
• Voice Compression Channels
SIP calls use system voice compression channels in the same way as used for standard IP
trunks and extensions. For an IP500 V2 system, these are provided by the installation of
VCM modules within the control unit. RTP relay is applied to SIP calls where applicable.
• Licensing
SIP trunks require licenses in the system configuration. These set the maximum number of
simultaneous SIP calls supported by the system.
• Firewall Traversal
Routing traditional H.323 VoIP calls through firewalls often fails due to the effects of NAT
(Network Address Translation). For SIP a number of ways to ensure successful firewall
traversal can be used. The system does not apply any firewall between LAN1 and LAN2 to
SIP calls.
- STUN (Simple Traverse of UDP NAT)
UDP SIP can use a mechanism called STUN to cross firewalls between the switch and
the ITSP. This requires the ITSP to provide the IP address of their STUN server and the
system to then select from various STUN methods how to connect to that server. The
system can attempt to auto-detect the required settings to successfully connect. To use
STUN, the line must be linked to the Network Topology settings of a LAN interface using
the line's Use Network Topology Info setting.
- TURN (Traversal Using Relay NAT)
TCP SIP can use a mechanism called TURN (Traversal Using Relay NAT). This is not
currently supported.
- Session Border Control
STUN does not have to be used for NAT traversal when SBC is between IP Office and the
ITSP, since the SBCE will be performing NAT traversal.
• SIP Trunks
These trunks are manually added to the system configuration. Typically a SIP trunk is
required for each SIP ITSP being used. The configuration provides methods for multiple
URI's from that ITSP to use the same trunk. For each trunk at least one SIP URI entry is
required, up to 150 SIP URI's are supported on the same trunk. Amongst other things this
sets the incoming and outgoing groups for call routing.
• Outgoing Call Routing
The initial routing uses any standard short code with a dial feature. The short code's Line
Group ID should be set to match the Outgoing Group ID of the SIP URI channels to use.
However the short code must also change the number dialed into a destination SIP URI

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SIP Trunk Overview

suitable for routing by the ITSP. In most cases, if the destination is a public telephone
network number, a URI of the form [email protected] is suitable. For example:
- Code: 9N#
- Feature: Dial
- Telephone Number: N"@example.com"
- Line Group ID: 100
While this can be done in the short code, it is not an absolute necessity. The ITSP Proxy
Address or ITSP Domain Name will be used as the host/domain part.
• Incoming Call Routing
Incoming SIP calls are routed in the same way as other incoming external calls. The caller
and called information in the SIP call header can be used to match Incoming CLI and
Incoming Number settings in normal system Incoming Call Route records.
• DiffServ Marking
DiffServ marking is applied to calls using the DiffServ Settings on the System > LAN > VoIP
tab of the LAN interface as set by the line's Use Network Topology Info setting.
SIP URIs
Calls across SIP require URI's (Uniform Resource Identifiers), one for the source and one for the
destination. Each SIP URI consists of two parts, the user part (for example name) and the domain
part (for example example.com) to form a full URI (in this case [email protected]). SIP URI's
can take several forms:
• [email protected]
• [email protected]
• [email protected]
Typically each account with a SIP service provider will include a SIP URI or a set of URI's. The
domain part is then used for the SIP trunk configured for routing calls to that provider. The user
part can be assigned either to an individual user if you have one URI per user for that ITSP, or it
can also be configured against the line for use by all users who have calls routed via that line.
Resource Limitation
A number of limits can affect the number of SIP calls. When one of these limits is reached
the following occurs: any further outgoing SIP calls are blocked unless some alternate route is
available using ARS; any incoming SIP calls are queued until the required resource becomes
available. Limiting factors are:
• the number of licensed SIP sessions.
• the number of SIP sessions configured for a SIP URI.
• the number of voice compression channels.
- SIP Line Call to/from Non-IP Devices Voice compression channel required.
- Outgoing SIP Line Call from IP Device No voice compression channel required.
- Incoming SIP Line Call to IP Device If using the same codec, voice compression channel
reserved until call connected. If using differing codecs then 2 channels used.

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 SIP Line Requirements

SIP Information Display

The full from and to SIP URI will be recorded for use by SMDR. For all other applications and
for telephone devices, the SIP URI is put through system directory matching the same as for
incoming CLI matching. First a match against the full URI is attempted, then a match against the
user part of the URI. Directory wildcards can also be used for the URI matching.
Related links
SIP Trunk Overview on page 831

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Chapter 84: SIP Headers and URIs

During SIP calls, various request and response messages are exchanged (see Request methods on
page 877 and Response methods on page 877). For example, a SIP call is started by the caller
sending an INVITE request to which 180 Ringing and 200 OK responses are expected.
These request and response messages contain various 'headers' detailing different information
values, see Headers on page 878. Some of these headers contain contact information in the
form of SIP URIs (Uniform Resource Identifier). For example; the caller, the original destination, the
current destination, and so on.
Related links
SIP URI Formats on page 836
Standard SIP Headers on page 837
Setting the SIP URI Host on page 837
Setting the SIP URI Content on page 838
Selecting the SIP Header Format Used on page 840

SIP URI Formats

When a header contains contact information, it is typically added using the 'SIP URI' format:
• A SIP URI is like an email address. In its simplest form, it appear as
sip:content@hostname.
• The SIP URI can also indicate the target port: sip:content@hostname:port
• For some headers, the SIP URI can also include preferred display name. When
that happens, the sip: part is enclosed in < > brackets. For example: display
<sip:content@hostname>.
- From the caller name, the line's Name Priority setting sets whether this name is displayed
on internal phones or replaced by a name match from the system or user directories.
• Some line providers may use a different format called TEL URI. That takes the form
tel:123456789.
• The system can use SIPS format which replaces sip: with sips:. This is used to indicate
that the connection should use TLS from end to end. That is, every hop should use TLS.
The header format used by each SIP line in the system configuration is set by its URI Type setting.
See Selecting the SIP Header Format Used on page 840.

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 Standard SIP Headers

Related links
SIP Headers and URIs on page 836

Standard SIP Headers

Most request messages exchanged during SIP calls include the following headers:
Header Description
Request-URI Also known as Request-Line-URI or R-URI. The first line of the request message
indicates the destination for routing the message.
If the message is routed through multiple hops, this header changes each time to
indicate the next destination. Similarly, if the call is redirected, the header is changed
to show the new destination.
Via Every intermediate proxy that has been involved in the routing of the request is included
as a SIP URI. These are used in reverse order to process response messages.
To This header indicates the original intended call target. It contains a SIP URI and can
include a display name.
• On outgoing calls, the identity of the called party is not known at the time of the initial
INVITE. Therefore the To: field only contains the information necessary to route the
call. That is, the dialed digits after any short code and prefix manipulation.
From This header indicates the identity of the caller. It contains a SIP URI and can include a
display name.
Contact This header indicates the return address for responses to the call request. This is a SIP
URI.
The SIP URI is similar to the From header. However, if anonymous calling
is enabled, the field becomes semi-anonymous. For example; Contact:
<sip:[email protected]:5060;transport=udp>
P-Asserted- The SIP URI is similar to the From header. The field is unchanged even if anonymous
Identity calling is enabled.

Related links
SIP Headers and URIs on page 836

Setting the SIP URI Host

When the system needs to send a SIP URI in a header, the information used for the @host part of
SIP URI is taken from the following settings (listed in order of priority, starting with the highest):

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SIP Headers and URIs

Source/Setting Descriptions
Short Code Short codes used to route calls to a SIP line can specify the host for the calls To and
R-URI headers.
• This is done in the short code's Telephone Number field by adding the host as a
quoted suffix. For example, N"@example.com".
• The value must be enclosed in " " quotation marks to prevent any part being
interpreted as short code wildcard characters.
Local Domain If set, this setting is used for the host part the From, Contact and Diversion headers
Name sent by the system, overriding the ITSP Domain Name below. It is also used for the PAI
header if Use Domain for PAI is selected on the SIP line.
ITSP Domain Name If set, this setting is used for the host part of the From, To, Diversion and R-URI
headers sent by the system.
ITSP Proxy This setting is used for the host part of most headers sent by the system if none of the
Address values above are set. However, if several addresses are set here, then either the ITSP
Domain Name and/or Local Domain Name settings must be used.

Related links
SIP Headers and URIs on page 836

Setting the SIP URI Content

Each SIP line in the system has a Call Details form which can contains SIP URI entries. They
set which headers are used and how the header data is populated or the values against which
headers are matched.
Every incoming and outgoing call which uses the line is matched to one of these.
Display
As above, the line URI associated with the call also sets the display name source for any headers
that require it, for example the From and To headers.
Setting Description
Auto The system automatically determines the appropriate value to use. It will use external
numbers if forwarding incoming calls, and internal numbers for calls made by a local
user.
• On incoming calls, if the Local URI is set to Auto, the system looks for matches
against extension numbers and system short codes.
• On outgoing calls, it allows short code manipulation of the caller number and name.
For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
Table continues…

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 Setting the SIP URI Content

Setting Description
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead.
Manual Entry If required, you can type in a value. This is only used for fields configured as Explicit.
(Explicit) This is typically used to set the DDI to be associated with SIP line appearances.
Credential Values If a set of SIP Credentials has been selected in the URI settings, then the User name,
Authentication Name or Contact values from the SIP credentials can be selected as
values.

Content
On both incoming and outgoing SIP calls, the system associates one of the SIP line's URI entries
with the call. The settings of that URI specify how the system should populate and use the
content part of the SIP URI in various header. The possible settings are:
Setting Description
Auto If Auto is selected, the system automatically determines the appropriate value to use. It
uses external numbers when forwarding incoming calls, and internal extension numbers
for calls made by a local user.
• On incoming calls, the system looks for matches against extension numbers and
system short codes.
• On outgoing calls, the system allows short code manipulation of the caller number and
name. For example: S to explicitly set the caller number, W to set withheld, A to allow
(override any previous withhold setting), Z to set the caller name.
Use Internal Data Use the SIP settings of the user (User > SIP), group (Group > SIP) or voicemail services
(System > Voicemail > SIP) making or receiving the call:
• Use the SIP Display Name (Alias) setting.
• If the Anonymous is selected, use that value instead. See Anonymous SIP Calls on
page 842.
Manual Entry If required, you can manually type in a value to use. The value is then used by other
fields configured as Explicit. This is typically used to set the DDI to be associated with
SIP line appearances.
Credential Values If a Credentials entry has been selected above, then the User name, Authentication
Name and Contact values from the selected credentials entry can be selected as values.
The value is then used by other fields configured as Explicit.
• URI values should only be set using credentials when required by the line provider.
For example, some line providers require the From header to always contains the
credentials used for registration, whilst other headers are used to convey information
about the caller ID.

Related links
SIP Headers and URIs on page 836

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SIP Headers and URIs

Selecting the SIP Header Format Used

The header format used by the system is set by the SIP Line > URI Type setting. This has the
following options:
• SIP - Use sip: format SIP URIs.
• Tel - When selected, the system uses the Tel URI format, for example tel:
+1-816-555-1212, in To headers.
• SIPS - When selected, the system replace the sip: part of the SIP URIs it sends with
sips:. That indicates that TLS must be used for all stages of the call. To use this, the line's
Layer 4 Protocol needs to be set to TLS.
Related links
SIP Headers and URIs on page 836

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Chapter 85: Outgoing SIP Call Routing

This section describes the overall processes used by the IP Office to route outgoing SIP trunk calls.
Related links
SIP Outgoing Call Routing on page 841
Anonymous SIP Calls on page 842
SIP ARS Response Codes on page 843
Typical outgoing call scenarios on page 845

SIP Outgoing Call Routing

When a user makes a call by dialing a number:
1. Dial Short Code Completed:
The dialing is processed though user, user rights, system and ARS short codes.
2. SIP Line URI/Line Appearance Matching:
The IP Office looks for a SIP URI with the same Outgoing Group as the short code's Line
Group ID.
a. Each line is checked in Line Number order.
b. SIP Line Appearance entries are checked first, then SIP URI entries.
c. Entries that have reached their Max Sessions or Outgoing Sessions are skipped.
d. Once a match is found, it and the SIP line to which it belongs are used:
• The line's Call Initiation Timeout (s) (default 4 seconds) sets how long the IP
Office waits for a response to the attempt to initiate a call before following any
alternate routes set in the ARS form.
• The line's Call Queuing Timeout (m) (default 5 minutes) sets how long the IP
Office waits for the call to be answered after receiving a provisional response.
3. No Available URI:
If all possible matches have reached their Max Sessions or Outgoing Sessions value:
• If the call was routed via an ARS short code, the ARS settings determine whether the
call can be redirected to an alternate route.
• Otherwise, the call waits for a matching URI to become available ("Waiting for
Line").

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Outgoing SIP Call Routing

Related links
Outgoing SIP Call Routing on page 841

Anonymous SIP Calls

Calls can be made and received with indication that the caller ID should be withheld. In SIP
terminology, these are 'anonymous' calls.
Important:
• Some line providers do not support the use of anonymous and will drop those calls.
Others may require additional configuration in order to accept the use of anonymous
status.
For outgoing calls, the call can be set as anonymous using the following methods:
• W Short Code Character
Adding a W as a suffix to the Telephone Number setting of a short code indicates withhold
the caller ID. For SIP calls, this is supported if the line URI being used is set to Auto.
• User/Group Anonymous Setting
Each system user and hunt group has a set of SIP settings (User > SIP, Group > SIP,
System > Voicemail > SIP ). If selected, the Anonymous setting indicates that the user
or group should treated as anonymous when making/receiving SIP calls. The voicemail
service also has SIP settings that include the option to be anonymous. For SIP calls, this is
supported if the line URI used is set to Use Internal Data.
• Withhold Number Option
Avaya feature phones can be configured to withhold the caller ID (Features > Call
Settings > Withhold Number). That matches the W short code operation above.
How Does Setting a Call as Anonymous Affect the Call Headers?
Setting anonymous/caller ID withheld has the following effect on the information added to SIP
URIs sent by the system:
• The system adds a Privacy header to the call information.
• If Send From In Clear is not enabled (the default):
- The SIP URI in the From header is anonymized:
• The display name part is set to "Anonymous".
• The content and host parts are set to dummy values
([email protected]).
- The Privacy header is set to user, otherwise it is set to id.
- A PPI or PAI header is used to contain the caller's number. This is done using the line's
Use PAI for Privacy setting (off and so PPI is used by default). PAI headers should only
be used in a trusted network.

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 SIP ARS Response Codes

• If Send From In Clear is enabled:

- The From header is not anonymized,
• When used in SIP URI and SIP Line Appearances, the P Preferred ID or P Asserted ID
entries should be configured to be the same or only one of them should be configured.
Related links
Outgoing SIP Call Routing on page 841

SIP ARS Response Codes

Through SIP RFC3398, many of the response codes used for SIP calls are translations of ISDN
codes. For outgoing calls, these can affect the routing through ARS as follows:
Do Not Use This Line Group
The following response codes will cause the system's ARS to no longer target the particular
outgoing line group. It depends on other settings whether the ARS attempts to target the call to a
different line group or escalate it to another ARS entry.
Code Cause Code
1 Unallocated Number.
2 No route to specific transit network/(5ESS) Calling party off hold.
3 No route to destination./(5ESS) Calling party dropped while on hold.
4 Send special information tone/(NI-2) Vacant Code.
5 Misdialed trunk prefix.
8 Preemption/(NI-2) Prefix 0 dialed in error.
9 Preemption, cct reserved/ (NI-2) Prefix 1 dialed in error.
10 (NI-2) Prefix 1 not dialed.
11 (NI-2) Excessive digits received call proceeding.
22 Number Changed.
28 Invalid Format Number.
29 Facility Rejected.
50 Requested Facility Not Subscribed.
52 Outgoing calls barred.
57 Bearer Capability Not Authorized.
63 Service or Option Unavailable.
65 Bearer Capability Not Implemented.
66 Channel Type Not Implemented.
69 Requested Facility Not Implemented.
Table continues…

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Outgoing SIP Call Routing

Code Cause Code

70 Only Restricted Digital Information Bearer Capability Is Available.
79 Service Or Option Not Implemented.
88 Incompatible.
91 Invalid Transit Network Selection.
95 Invalid Message.
96 Missing Mandatory IE.
97 Message Type Nonexistent Or Not Implemented.
98 Message Not Implemented.
99 Parameter Not Implemented.
100 Invalid IE Contents.
101 Msg Not Compatible.
111 Protocol Error.
127 Interworking Unspecified.

Stop ARS
The following response codes end the outgoing call routing and any further ARS targeting of the
call.
Code Cause Code
17 Busy.
21 Call Rejected.
27 Destination Out of Order.

No Affect
All other cause codes do not affect ARS operation.
Related links
Outgoing SIP Call Routing on page 841

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 Typical outgoing call scenarios

Typical outgoing call scenarios

INVITE with SDP, local ringback

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Outgoing SIP Call Routing

INVITE with SDP, early media

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 Typical outgoing call scenarios

INVITE with SDP, early media re-directed by destination

Related links
Outgoing SIP Call Routing on page 841

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Chapter 86: Incoming SIP Call Routing

This section describes the overall processes used by the IP Office to route incoming SIP trunk calls.
Related links
SIP Short Codes on page 848
SIP Incoming Call Routing on page 849
SIP Prefix Operation on page 851
Media path connection on page 851
SIP Caller Name and Number Display on page 852
Typical incoming call scenarios on page 853

SIP Short Codes

Outgoing SIP calls are largely processed through short codes in the same way as for other line
types. The following specific notes apply:
Note Description
En-Bloc Dialing is SIP Lines do not use overlap dialing. They expect to receive the full destination number,
Required called 'en-bloc' dialing.
• The short code used to route calls to a SIP Line should use a ; (semi-colon) character
at the end of the short code field. That character instructions the system to wait for
dialing to be completed before using the short code.
• Dialing complete is indicated by either:
- the dialer pressing #
- the device or application being used sending a dialing complete signal.
- the IP Office's Dial Delay Time expiring. The default is 4 seconds.
Caller ID For SIP URIs configured to Auto (the default), the short code Telephone Number field
Characters characters used to control the sending of the caller ID number are supported. Those
characters are A, W, S and SS.
Host ID Short codes used to route the call to a SIP line can specify the host to be used for To
and R-URI headers. This is done in the short code's Telephone Number field by adding
the host as a quoted suffix.
For example, N"@example.com". The value must be enclosed in " " quotation marks
in order to prevent its characters being interpreted as short code wildcards.

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 SIP Incoming Call Routing

Related links
Incoming SIP Call Routing on page 848

SIP Incoming Call Routing

When the IP Office receives a SIP call, it determines the routing of the call as follows:
1. Line Matching:
The incoming SIP call is matched to a SIP line. If no match is found, the call is ignored. SIP
line matching is done in two stages:
a. Protocol Matching:
The call is matched with lines configured with the same protocol (UDP, TCP or TLS)
and listening port settings.
b. Line Association:
If there are several possible line matches, they are checked in Line Number order,
for a match between the incoming call's source and each line's Association Method
setting.
2. Line Call Details Matching:
Using the line's Call Routing Method, either the To or Request-URI of the incoming SIP
request is used to find a matching Local URI.
• The IP Office first looks for a matching SIP Line Appearances. If a match is found, the
call is associated with the first available line appearance number in the Incoming ID
order setting.
• If no SIP Line Appearances match is found, the IP Office checks the line's SIP URIs in
their URI number order.
• Entries at their Max Session or Incoming Sessions values are ignored.
• When an match is found, its Incoming Group ID setting is used for incoming call route
matching. For SIP Line Appearances, this is in addition to altering on matching line
appearance buttons.
• If no match is found, the IP Office uses its Service Busy Response setting (486 Busy
here or 503 Service Unavailable) to end the call.
3. Incoming Call Route matching:
The IP Office's incoming call route entries are checked for a match.
a. The matching uses the following options in order:
a. Line Group Match:
Only incoming call routes with a Line Group ID setting that matches the SIP
lines appearance/SIP URI's Incoming Group ID setting are checked.
b. Incoming Number Match:
The IP Office looks for call routes with a match between their Incoming Number
setting and the Local URI value received. There is always a received number

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Incoming SIP Call Routing

value with incoming SIP calls, so always a potentially incoming number matching
value.
• Incoming calls routes with a blank Incoming Number field match any
incoming number.
• If the incoming call route's Destination is set to . (period), the Local URI
received is used to look for destination matches.
- If set to Auto, the IP Office looks for a matching extension number or
system short code.
- If set to Use Internal Data, the system looks for a match against the SIP
Name of users and then groups.
c. Incoming CLI Match
From the possible matches, the IP Office looks for a match between the each
route's Incoming CLI, if set, and the caller details in the From header. For SIP
URI and TEL URI headers, partial matching starting from the left is supported.
For IP addresses, only exact matches are supported.
b. If the call matches more than one incoming call route:
a. The most precise match is used. For example, the highest number of matching
criteria and highest number of exact digit rather than wildcard character matches.
b. If the call stills matches more than one incoming call route, the one added to the
configuration first is used.
c. If there is no match:
a. For calls using a line's SIP URI entry with its Local URI set to Auto, the incoming
number is checked for a direct match to an internal extension number.
b. Otherwise, busy indication is sent to the caller and the call is dropped.
4. Incoming Call Route Match:
Once a match is resolved, this determines the incoming call route’s current destination:
a. Each incoming route can include multiple pairs of main and fallback destinations.
b. Apart from the default pair, each pair uses an associated time profile. The time profile
defines when that destination pair should be used.
a. With multiple destination pairs, the entry used is the first, working from bottom
up, whose time profile is currently 'true'. If no match occurs, the Default Value
options are used.
b. The system attempts to present the call to the destination. If the destination is
busy, it presents the call to the fallback extension.
5. Call Presentation:
The call is presented to the destination. If the call was routed via a SIP Line Appearance,
the call also alerts on any matching Line Appearance buttons.
Related links
Incoming SIP Call Routing on page 848

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 SIP Prefix Operation

SIP Prefix Operation

The SIP line settings include settings for Prefix, National Prefix, Country Code and
International Prefix values. These values are used in the following order:
1. If the number starts with a + symbol, the symbol is replaced with the International Prefix.
2. If the Country Code has been set:
a. If the number begins with the Country Code, or International Prefix plus Country
Code, the IP Office replaces them with the National Prefix.
b. If the number does not start with the National Prefix or International Prefix, the IP
Office adds the International Prefix.
3. If the incoming number does not begin with the National Prefix or International Prefix,
the IP Office adds the Prefix.
Examples
For example, if the SIP Line is configured with prefixes as follows:
• Prefix: 9 - The external dialing prefix used to make outgoing external calls.
• National Prefix: 90 - The expected prefix for outgoing national calls, including the external
dialing prefix.
• International Prefix: 900 - The expected prefix for outgoing international calls, including the
external dialing prefix.
• Country Code: 44 - The local country code.
Number Received Processing Resulting Number
+441707362200 Following rule 1, the + is replace with the International 901707362200
Prefix.
The number now matches the International Prefix and
Country Code. Following rule 2a, they are replace with the
National Prefix.
00441707362200 Following rule 2a, the International Prefix and the Country 90107362200
Code are replaced with the National Prefix.
441707362200 Following rule 2a, the Country Code is replace with the 901707362200
National Prefix.
6494770557 Following rule 3, the International Prefix (900) is added. 9006494770557

Related links
Incoming SIP Call Routing on page 848

Media path connection

IP Office does not provide in-band ringback to incoming SIP trunk calls. The only normal scenario
in which an incoming SIP trunk call hears in-band ringback occurs when the call terminates on an

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Incoming SIP Call Routing

analog trunk. With analog trunks, the media path is cut through immediately because IP Office has
no way of determining the state (ringing, busy, answered) of the trunk.
IP Office can connect “early” media before the call is answered by sending a 183 Session
Progress response. This is only done when the following two conditions are met:
• A PROGRESS (in-band tone indication or 183 Session Progress with SDP) message is
received from the destination. This can only happen in a SIP-to-PRI or SIP-to-SIP tandem
call scenario.
• The INVITE message contains SDP.
- IP Office does not attempt to connect early media on PROGRESS when there is no SDP
in the initial INVITE, since this is unlikely to succeed. The likely reason there is no SDP
in the INVITE is probably that the originating system does not know the originator’s
media address yet. A typical scenario where this is the case occurs when the call on the
originating system comes from an H.323 SlowStart trunk.
Related links
Incoming SIP Call Routing on page 848

SIP Caller Name and Number Display

For incoming SIP calls, the caller name and number are obtained from the following headers:
Value Description
Caller Name Unless withheld (see notes below), the caller number for incoming calls is taken from
PAI header if present, otherwise from the From header. The SIP line's advanced settings
Caller ID FROM Header option can be used to force use of the FROM header only.
Caller Number The caller name for incoming calls is taken from the name supplied with the following
headers, in order of priority with highest first:
1. PPI header
2. PAI header
3. Remote Party ID header
4. Contact header

Notes
1. The above apply regardless of the header settings of the SIP URI handing the incoming
call. For example, for incoming caller details, you do not need to have P Preferred ID
selected and configured in the SIP URI or SIP line appearance. The PPI header info is
used if present in the incoming request.
2. If the receiving IP Office system has Caller ID from From header enabled (disabled by
default), then the From header name is used regardless of PAI or PPI headers.
3. If the header to be used for the caller’s name does not contain a name, “Unknown” is
displayed.

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 Typical incoming call scenarios

4. Calls from a source that is anonymous display "Withheld" as the caller name and no
number.
Related links
Incoming SIP Call Routing on page 848

Typical incoming call scenarios

INVITE with SDP, local ringback
If the destination is an analog trunk, the 180 Ringing will be replaced with a 183 Progress with
SDP followed immediately by a “fake” answer in order that the media will be connected right away
so that the originator hears whatever in-band tones are present on the analog trunk (ringback or
busy). If the target is an extension that is unconditionally call forwarded over an analog trunk, then
there will be a 180 Ringing without SDP, followed immediately by the “fake” answer.

INVITE with SDP, early media

If the SIP Trunk receives a FAR_PROGRESS (in-band) message from its peer in the core (e.g.
from a tandem PRI or SIP trunk), it sends a 183 Session Progress message with SDP to the far
end. IP Office will connect the media on receipt of 180 or 183 with SDP.

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Incoming SIP Call Routing

INVITE without SDP, local ring back

IP Office does not attempt to send early media in this scenario.

INVITE without SDP, early media

In this scenario, the far end attempts to connect media before the call is answered. IP Office does
not provide early media when receiving an empty INVITE, but rather 180 Ringing instead. There is

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 Typical incoming call scenarios

no requirement to provide an SDP in the 180 Ringing provisional response, as that response is not
sent reliably using the PRACK mechanism.

INVITE without SDP, call terminates on H.323 endpoint

If the destination of the call is an H.323 trunk, the destination media address is not known when
the call is answered. Therefore, the SDP offer in 200 OK will contain a null port number (and IP
address). Once the logical channels are opened on the H.323 side, IP Office sends a re-INVITE
using the real media address.

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Incoming SIP Call Routing

Related links
Incoming SIP Call Routing on page 848

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Chapter 87: SIP messaging

SIP trunk prerequisites

Before any calls can be made, the system must have sufficient SIP trunk licenses for the maximum
number of simultaneous SIP trunk calls expected.
On Server Edition systems, the System | Telephony | Telephony | Maximum SIP Sessions value
must match the total number of SIP extension and trunk calls that can occur at the same time.
Related links
Codec selection on page 857
SIP DTMF Transmission on page 858
Fax over SIP on page 859
SIP Call Hold Scenarios on page 859
SIP Call Transfers (Refer) on page 861
Ringback Tone on page 862
Hold Reminders on page 863

Codec selection
Normal Codec Selection
Codec selection is based on the Offer/Answer model specified in RFC 3264.
1. The calling endpoint issues an offer that includes a list of the codecs it supports.
• For IP Office SIP trunks, the IP Office offers the codecs set on the SIP trunks VoIP tab.
It does not offer those set on the extension.
2. The called endpoint sends an answer that normally contains a single codec from the
offered list.
• If there are multiple codecs in the answer, IP Office only considers the first codec. If the
SIP Line is configured to do Codec Lockdown, it will send another INVITE with the
single chosen codec.
Codec Changes with reINVITE
For R11.0 and higher, the IP Office supports codec selection following a reINVITE. Previously,
when a reINVITE was received during a call, if the reINVITE contained the codec currently in

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SIP messaging

use, that codec was preferred and kept. For R11.0 and higher, the IP Office reevaluates the codec
to use based on any preferences included in the reINVITE:
• For example, if the endpoint/trunk has a different codec preference to the system, hold/
unhold sequences will result in codec changes. When held, the system codec preference is
used to play music-on-hold. When unheld, the codec preferences are reevaluated.
When using this behavior:
• Direct media is supported for SRTP phones that change keys on each reINVITE.
• The IP Office supports the transfer of video calls.
Note:
• The new behavior also applies to SM lines and SIP extensions.
• On IP Office systems upgraded to R11.0 and higher, SLIC_PREFER_EXISTING_CODEC
is automatically added to the SIP Engineering tab of any existing SIP lines to retain the
existing pre-R11.0 behavior.
Related links
SIP messaging on page 857

SIP DTMF Transmission

DTMF key presses can be transmitted either in-band as audio tones or signaled using DTMF over
RTP (RFC 2833)
When using DTMF over RTP (RFC 2833), the IP Office supports asymmetric dynamic payload
negotiation when it is necessary to bridge SIP endpoints that do not support payload negotiation.
The value used for an initial offer is configured on the System | Codecs tab. The default value
is 101. Upon receipt of an offer with an RFC2833 payload type, IP Office automatically uses the
proposed value rather than its own configured value. This helps to support networks that do not
negotiate payload types.
Direct Media Calls
There are cases in which direct media is desirable between SIP trunks and endpoints that do
not support RFC2833. To allow for this, if a key presses is indicate from the extension, the IP
Office temporarily switches the call back to indirect media. It then injects the digits in-band using
the negotiated dynamic payload and, after fifteen seconds of no further key presses, the call is
switched back to direct media.
Related links
SIP messaging on page 857

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 Fax over SIP

Fax over SIP

T.38 Fax over SIP is supported on the IP500 V2 platform deployed as standalone or as an
expansion gateway. G.711 fax is also supported, and is supported on Linux servers. For networks
that do or do not support T.38, IP Office allows both G3 and Super G3 fax machines to
interoperate.
There are configuration parameters that control the behavior in different networks. If T.38 is
supported in a network, then it may make sense to select T.38 as the Fax Transport preference
in order to make use the inherent quality provided by the redundancy mechanisms. On the other
hand, if all of the fax machines in the network are Super G3 capable, there may be a need to
take advantage of the increased speed that this encoding provides. Since T.38 is not capable of
encoding Super G3, G.711 may be a better choice for the Fax Transport. In either case, IP Office
will accept codec change requests from the far SIP endpoint to switch to T.38 or to G.711.
T.38 Fax Transport and Direct Media are mutually exclusive on a given SIP Line. IP Office keeps
itself in the media path so that it can detect fax tones to make the switch to T.38.
Related links
SIP messaging on page 857

SIP Call Hold Scenarios

Hold originated by IP Office
When an IP Office DS extension or non-IP trunk puts a SIP trunk on hold, there is no indication
to the network. The voice path is merely switched in the TDM domain to the appropriate hold
treatment source (tones, silence or music).
For IP extensions and trunks, be they H.323 or SIP, if the call uses direct media, there will be a
re-INVITE sent to redirect the media source from the extension or trunk endpoint to a port on the
IP Office in order to connect hold treatment. When the call is then unheld, another INVITE will go
out to connect the extension with the far end.

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SIP messaging

Hold originated by far end

The far end of a SIP trunk can put the IP Office call on hold by sending it an re-INVITE with an
SDP Offer containing:
• A sendonly attribute. IP Office replies with an SDP Answer containing the recvonly
attribute.
• An inactive attribute. IP Office replies with inactive.
• A zero media connection address (c=0.0.0.0). IP Office replies with inactive.
Unhold
A held call is unheld by means of an SDP Offer with the sendrecv attribute (or no direction
attribute, since sendrecv is assumed if not specified).
Unhold from mutual hold
Either end can unhold the other end by sending a new Offer with the sendrecv or
recvonlyattribute. The other end replies with sendonly if the call is still on hold at its end.
Related links
SIP messaging on page 857

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 SIP Call Transfers (Refer)

SIP Call Transfers (Refer)

After a SIP call has been established, the SIP REFER method is used by the transferor end of
the call to transfer the transferee end to a the transfer target. The REFER message provides the
transfer target’s contact information in a Refer-To header. That information is used to establish
complete the transfer.
For public SIP trunks, IP Office supports only consultative call transfer using REFER. Consultative
transfers are also known as attended or supervised transfers. With consultative transfer, the
transferor puts the first call on hold and establishes a consultation call to the transfer target. After
the consultation call, the transferor completes the transfer, causing the transferee to connect to the
transfer target, replacing the transferor.
REFER can be configured to accept incoming, reject incoming, or decide based on the presence
of REFER in the Allow: header in responses to OPTIONS messages. Similarly, there is the same
configuration for outgoing REFER.
Although the transferor and transferee must be SIP endpoints, the transfer target can be a TDM,
PRI, H.323 or SIP terminal on the same IP Office, or an endpoint reachable over the same SIP
line as the REFER request is received from.

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SIP messaging

Related links
SIP messaging on page 857

Ringback Tone
The ringback tone behavior of IP Office systems has changed for IP Office R11.0 and higher.
After sending an INVITE request, if the IP Office receives an 18X response with SDP, it starts
playing remote ringback tone. Prior to R11.0, if it then receives an 18X response without SDP, the
IP Office would continue playing remote ringback tone. For R11.0 and higher, following the 18X
without SDP, the IP Office now switches to local ringback tone.

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 Hold Reminders

In summary:
1. The IP Office sends an INVITE.
2. The IP Office receives 18X with SDP. The IP Office plays remote ringback tone.
3. The IP Office receives 18X without SDP:
• Pre-R11.0: Continue playing remote ringback tone.
• R11.0+: Switch to playing local ringback tone.
This feature is supported regardless of whether provisional response reliability (PRACK/100rel) is
enabled or not.
When SIP call signaling transitions from remote to local ringback, the IP Office hosting the SIP
trunk play the local ringback tone to the other end (phone or trunk).
Ringback Tone with Early Media
A special case applies for SIP trunks configured to use p-early-media. For 18x responses with
or without SDP to be considered, a p-early-media header must be present in the response.
If otherwise, the message is not considered with regards to early media (the system continues
playing either local ringback or remote early media).
For example: The IP Office receives a 183 response with SDP and a p-early-media header
with a sendonly or sendrecv parameter. The IP Office then receives a 183 response (with or
without SDP):
• Example 1: If the response does not include a p-early-media header, the IP Office
continues listening to the remote early media.
• Example2: If the response includes a p-early-media header with an inactive parameter,
the IP Office switches to playing local ringback tone.
Related links
SIP messaging on page 857

Hold Reminders
For IP Office R11.0 and higher:
• For SIP phones, the IP Office only provides hold reminders to Avaya SIP phones.
• If the user is on the video call, there will be no reminder call.
• The IP Office supports direct media when using SRTP with 1100, 1200, J129, E129, B179
and H175.
Related links
SIP messaging on page 857

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Chapter 88: SIP Line Appearances

The system can implement some degree of line appearance emulation on SIP trunks. Note the word
'emulation'.
Related links
SIP Line Appearance Incoming Call Routing on page 864
SIP Line Appearance Outgoing Call Routing on page 864
SIP Line Appearance User Button Programming on page 865

SIP Line Appearance Incoming Call Routing

The routing of incoming SIP calls, including SIP line appearances, is covered in SIP Incoming Call
Routing on page 849. However, the following key points should be observed:
• Call matching to an incoming call route destination (or an extension match to the Local URI
value) is still required. The call is rejected if that does not occur.
- This is necessary to associate the call with a user or group whose settings (for example
forwarding and voicemail) it follows until answered.
- If the incoming call route destination is a user with a matching line appearance button for
the call, then the additional private line features for a line appearance call are applied. See
Line Appearance Buttons on page 1176.
• As normal, if the call is answered by the voicemail service, whilst it is indicated on line
appearance buttons, they cannot be used to answer or bridge into the call.
Related links
SIP Line Appearances on page 864

SIP Line Appearance Outgoing Call Routing

The SIP line appearance entries can be access for outgoing calls in two ways:

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 SIP Line Appearance User Button Programming

Method Description
Short Code If the Line Group ID of a Dial short code matches the Outgoing Group ID of the SIP
Routing line appearance entry, with available outgoing sessions, then that SIP line appearance
can potentially be used as a match for outgoing SIP calls. See Outgoing SIP Call
Routing on page 841.
• SIP Line Appearances matches are used before SIP URI entries.
• This allows the SIP line appearance entries to be used by any user routed to that short
code. They do not need to have an programmed Line Appearance buttons available.
• For users without programmed line appearance buttons to also receive calls from the
SIP line appearance, they need to be targeted by its matching incoming call route.
Line Appearance For users with Line Appearance buttons programmed to the particular Line Appearance
Buttons ID numbers being used, they can initiate outgoing calls by pressing any idle line
appearance button (pressing a button that is in use, will potentially bridge into that call
unless it is connected to voicemail).
• The users dialing is still processed through short code matching. This allows normal
short code manipulation of the outgoing number and/or barring of selected numbers.
• The short code used to route calls to a SIP Line should use a ; (semi-colon) character
at the end of the short code field. That character instructions the system to wait for
dialing to be completed before using the short code. Dialing complete is indicated by
either:
- the dialer pressing #.
- the device/application being used sending a dialing complete signal.
- the system's Dial Delay Time expiring.
• In this scenario, the Line Group ID of the short code needs to match the Outgoing
Group of the SIP line appearance entry.

Related links
SIP Line Appearances on page 864

SIP Line Appearance User Button Programming

Line appearances buttons for SIP line appearances are programmed in the same way as for any
type of line appearance. However, the following additional requirement applies:
• Users with line appearance buttons for a particular SIP Line appearance entry must be
assigned buttons for all line appearance numbers allocated to that entry. By default that is 3
line appearance numbers.
Related links
SIP Line Appearances on page 864

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Chapter 89: SIP Calling Number Verification
(STIR/SHAKEN)

Calling number verification is a SIP feature where the calling number is verified by the ISP and the
results of that verification is included with the incoming call. The aim of this is to help reduce call
spoofing.
• Support for and use of SIP calling number verification is mandated by law for US/Canadian
locales. However, the feature can be enabled in any locale if supported by the local SIP ISP.
• This feature only does calling number verification. The display name information supplied with
calls is not verified.
Verification is done by the ITSP by looking at several factors:
• Is the calling number associated with the subscriber making the call?
• Is the call coming from a known customer?
• Is the call originated by the known ITSP?
• Was the call digitally signed and was the ITSP able to fetch the public certificate of the
originating service provider in order to verify that the SIP INVITE has not be changed during
transit.
The result of the verification process is then indicated in the call's headers using a verstat value:
• TN-Validation-Passed plus an attestation level (see the table below). For example, TN-
Validation-Passed-A.
• TN-Validation-Failed plus an attestation level (see the table below). For example, TN-
Validation-Failed-A.
• No-TN-Validation -
The attestation levels are:
Attestation Level Description
A Full Attestation The customer is known and the calling number is one associated with that
customer.
• Note that for calls where no authentication level is indicated or can be
obtained, the IP Office treats the call as attestation level A.
Table continues…

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 The STIR/SHAKEN SIP Protocols

Attestation Level Description

B Partial Attestation The customer is known. However, the number not one associated with
that customer. For example:
• the customer is forwarding a call from with a original calling number that
is not associated with them.
• the call is originating from another known ITSP. Common for
international calls.
C Gateway Attestation The call has come via a trusted source, but the original customer and
number are not known.

When calling number verification is available, the IP Office system can use the results to determine
how to handle calls.
• Use of calling number verification is enabled on a per line basis.
• On lines where it is enabled, the line can either use the system default settings or line specific
settings
• The settings determine whether a call should accepted or not.
- If not accepted, the call is rejected by the system with a 666 response code.
- If accepted, the call is routed as normal by features such as Incoming Call Route matching.
However, if required, the specific result of the calling number verification can be used to vary
the routing.
• The attestation level is included in the call's SMDR record. That includes rejected calls.
Related links
The STIR/SHAKEN SIP Protocols on page 867
Obtaining a call's number verification result on page 868
Setting the system's number verification default behavior on page 868
Enabling calling number verification on a SIP line on page 869
SIP Calling Number Verification (STIR/SHAKEN) on page 870
Changing the rejected call responses on page 872
Changing the authentication header used on page 872
Customizing the call handling behavior on page 873
Call Records on page 873

The STIR/SHAKEN SIP Protocols

Calling number verification is implemented by ITSPs using a number of SIP RFCs, collectively
referred to as STIR/SHAKEN.
• STIR (Secure Telephony Identity Revisited)
- This protocol uses digital certificates between the customer (the call originator) and the
ITSP to establish customer authentication. The ISP can then examine known numbers
allocated to that customer for number authentication.

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SIP Calling Number Verification (STIR/SHAKEN)

• SHAKEN (Signature-base Handling of Asserted information using toKENs)

- These are guidelines for PSTN network providers handling calls which transit from the
non-SIP PSTN to SIP networks. Currently, it has mainly been implemented as a service for
SS7 carriers in USA and Canada.
For more details, refer to https://en.wikipedia.org/wiki/STIR/SHAKEN.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Obtaining a call's number verification result

The methods implemented by different ITSPs to send a call's number verification can vary. The
method used by IP Office to obtain the result is as follows:
• The IP Office looks for a verstat parameter in the tel or sip uri included in the call's
From or PAI header. If both are present, preference is given to the PAI header. For example:
- PAI: tel:+123456789;verstat=TN-Validation-Passed-A
- PAI: sip:+123456789;verstat=TN-Validation-Passed-
[email protected];user=phone
• The verstat parameter is used even if associated with an anonymous sip/sips URI.
• If there are multiple PAI headers with verstat information, only the first one is used.
• The attestation level is taken from the verstat parameter if it contains one of the
recognized authentication levels A, B and C.
• If the verstat parameter is not available, the IP Office checks whether the level is available
in any other headers such as Attestation-Info (used by ASBCE) or X-Attestation-Info (used
by Verizon).
• The checking of further headers can also be enabled. See Changing the authentication
header used on page 872.
• If no header provides an attestation level, level A is assumed.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Setting the system's number verification default behavior

This process sets the default behaviors applied by SIP lines on which calling number verification is
enabled.

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 Enabling calling number verification on a SIP line

Procedure
1. Access the System > VoIP > VoIP Security settings.
2. In the Calling Number Verification section, set the required behavior:
Field Description
Incoming Calls Default = Allow Not Failed
Handling
Sets the defaults for which calls are accepted by the system based on the authentication
level of the call. This default can be overridden in the individual line configuration.
• Allow All - Allow all calls regardless of calling number verification.
• Allow Validated - Only accept verified calls with full or partial attestation.
• Allow Not Failed - Accept all calls expect those that specifically failed verification.
Note this can include calls with no reported verification result.
Validation Default = Off
Presentation
If enabled, the system will prefix the caller ID information displayed on phones with a
character indicating the result of the call's validation result. This will be:
• A tick mark for full verification.
• A question mark for partial verification.
• A cross for authentication failed.
When enabled, the system will also inspect the display information on all received trunk
calls to ensure they do not start with these characters in order to avoid spoofing.

3. Save the settings.

Next steps
• Enable calling number validation on the individual SIP lines. See Enabling calling number
verification on a SIP line on page 869.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Enabling calling number verification on a SIP line

This process configures the SIP line specific settings for calling number verification.
Procedure
1. Access the SIP line's settings and select the SIP Advanced tab.
2. In the Calling Number Verification section, set the required behavior:

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SIP Calling Number Verification (STIR/SHAKEN)

Field Description
Calling Number Default = Off
Verification
Sets whether the line uses calling number verification.
Incoming Calls Handling Default = Allow Not Failed
Set which calls are accepted by the system based on the attestation level of the
call.
• System - Use the default system setting (System VoIP > VoIP Security >
Callng Number Verification).
• Allow All - Allow all calls regardless of calling number verification.
• Allow Validated - Only accept verified calls with full or partial attestation.
• Allow Not Failed - Accept all calls expect those that specifically failed
verification. Note this can include calls with no reported verification result.

3. Save the changes.

Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

SIP Calling Number Verification (STIR/SHAKEN)

For calls with are allowed following calling number verification, normal incoming call routing is
applied. However, that routing can be made to be specific to the verification result and attestation
level of the call.
This is achieved using the following characters in the Code field of short codes or Incoming CLI
field of incoming call routes:
Character Meaning Description
P Passed Matches calls where the verstat value is set to TN-Validation-Passed plus
the attestation level. For example, TN-Validation-Passed-A.
If required, the specific level of attestation to match can be specified. That is done
by following the P character with the required level or levels inside " " marks. For
example:
• P"A" matches calls with an attestation of A.
• P"B" matches calls with an attestation of B.
• P"AB" matches calls with an attestation of A or B.
F Failed Matches calls that specifically failed verification. That is, the call's verstat value
is set to TN-Validation-Failed.
Q Unknown Matches calls which do not have any verification result or where the verstat
value received is No-TN-Validation.

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 SIP Calling Number Verification (STIR/SHAKEN)

Example Incoming Call Routing

In this example, the system has the following incoming call routes are defined for calls to the
business's main sales number. Apart from the setting below, each of the incoming call routes has
the same settings and matches the incoming group ID used by the SIP trunks URI.
Incoming Call Incoming CLI Destination Description
Route
1. P"A" Hunt Group The business's sales hunt group.
2. blank Auto-Attendant The business's auto-attendant. The use of an auto-
3. P"B" Auto-Attendant attendant to answer calls deflects automated calls with
potential spoofed caller ID numbers but still allows
4. Q Auto-Attendant callers to select to be connected to the sales group
via the provided auto-attendant options.
5. F Barred A short code set to the barred feature.

The following calls to the sales number are received:

Incoming Call Incoming CLI Attestation Destination
Details Level
1. 111 A A fully validated call. The call details match the 2nd
call route only. Therefore, the call is routed the sales
hunt group.
2. 222 B A partially validated call. The call details match both
the 2nd and 3nd call routes. However, the 3rd match
is more precise, so the call is routed to the auto-
attendant.
3. 333 C The call details match both the 2nd and 5th call routes.
However, the 5th match is more precise, so the call is
routed to the barred short code.
4. 444 None A call with no attestation level. The call details match
the 2nd and 4th call routes. However, the 3rd match
is more precise, so the call is routed to the auto-
attendant.

Calling Name Display

Calling number validation only validates the caller's number, not the display name information
provided with the call.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

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SIP Calling Number Verification (STIR/SHAKEN)

Changing the rejected call responses

For rejected calls, by default the rejection is done using the response code 666 and the string
"Unwanted". However, if required by the ISP, a different code and/or string can be used,
configured using the process below.
• Rejected calls are included in the system's SMDR and CDR outputs.
• Rejected calls are not included in the individual user's call logs and call histories.
Procedure
1. Open the SIP line's settings and select SIP Engineering.
2. Click Add and enter one of the following custom strings:
• To change the reject code, enter SLIC_STIR_REJECT_CODE=N where N is the
response code number to use.
• To change the reject string, enter SLIC_STIR_REJECT_STRING=Y where Y is the string
to use.
3. Click Create new.
4. Save the settings.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Changing the authentication header used

Obtaining a call's number verification result on page 868 describes the normal process by which
the IP Office system normally obtains a call's verification result from its headers. However, if
required, the IP Office can look for the verstat value in another specified header.
• As per normal operation, if verstat values are present in more than one header, only the
first is used.
Procedure
1. Open the SIP line's settings and select SIP Engineering.
2. Click Add and enter one of the following custom strings:
• To specify the header to check, enter SLIC_STIR_ATTEST="W" where W is the name of
the header used by the ITSP.
- For example, SLIC_STIR_ATTEST="X-StirResult" instructs the IP Office to also
check for a value in the X-StirResult header if present. .
3. Click Create new.

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 Customizing the call handling behavior

4. Save the settings.

Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Customizing the call handling behavior

The behavior applied to calls can be customized. This is done on a per-line basis, using the
decimal sum of a binary bit string, where bit 0 is least significant bit (right-to-left).
The custom behavior provided by each bit when enabled (set to 1) are:
Bit Attestation Level Custom Behavior if bit set to 1
0 Attestation Passed Calls (A and B) Retain the Caller ID display.
1 Perform directory matching.
2 No Attestation Calls (Assumed A) Retain the Caller ID display.
3 Perform directory matching.
4 Attestation Failed Calls (C) Retain the Caller ID display
5 Perform directory matching.

Procedure
1. Open the SIP line's settings and select SIP Engineering.
2. Click Add and enter one of the following custom strings:
• To change the reject code, enter SLIC_STIR_CUSTOM=Z where Z is the decimal sum of
the binary bits.
- For example, SLIC_STIR_CUSTOM=15 retains the caller ID display and does
directory matching for all calls expect those that have attestation level C. That is,
bits 0 to 3 all set to 1, bits 4 and 5 set to 0. The decimal sum of that bit string is 15.
3. Click Create new.
4. Save the settings.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

Call Records
The authentication level (A, B or C) provided by the ISP is included in the SMDR call logging
records output by the system. If no authentication level is provided, N/A is shown instead.

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SIP Calling Number Verification (STIR/SHAKEN)

An SMDR call record is produced even for calls which are rejected by the system based on the
calling number verification settings.
Related links
SIP Calling Number Verification (STIR/SHAKEN) on page 866

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Chapter 90: IP Office SIP trunk
specifications

This section outlines the SIP trunk capabilities supported by IP Office.

Related links
SIP RFCs on page 875
Transport protocols on page 877
Request methods on page 877
Response methods on page 877
Headers on page 878

SIP RFCs
IP Office supports the following SIP RFCs:
RFC Title
– ITU-T T.38 Annex D, Procedures for real-time Group 3 facsimile communication over IP
networks
1889 RTP: A Transport Protocol for Real-Time Applications
2327 SDP: Session Description Protocol
2617 HTTP Authentication: Basic and Digest Access Authentication
2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
2976 The SIP INFO Method
3087 Control of Service Context using SIP Request-URI
3261 Session Initiation Protocol
3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
3263 Session Initiation Protocol (SIP): Locating SIP Servers
3264 An Offer/Answer Model with the Session Description Protocol (SDP)
3311 The Session Initiation Protocol (SIP) UPDATE Method
3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)
Table continues…

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IP Office SIP trunk specifications

RFC Title
3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted
3326 The Reason Header Field for the Session Initiation Protocol (SIP)
3329 Security Mechanism Agreement for the Session Initiation Protocol (SIP)
3398 Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP)
Mapping
3407 Session Description Protocol (SDP) Simple Capability
3489 STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address
Translators (NATs)
3515 The Session Initiation Protocol (SIP) Refer method
3550 RTP: A Transport Protocol for Real-Time Applications
3551 RTP Profile for Audio and Video Conferences with Minimal Control
3665 Session Initiation Protocol Basic Call Flow Examples
3666 Session Initiation Protocol PSTN Call Flows
3725 Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)
3824 Using E.164 numbers with the Session Initiation Protocol (SIP)
3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation
Protocol
3891 The Session Initiation Protocol (SIP) "Replaces" Header
3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)
4028 Session Timers in the Session Initiation Protocol (SIP)
4119 A Presence-based GEOPRIV Location Object Format
4566 SDP: Session Description Protocol
4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
5139 Revised Civic Location Format for Presence Information Data Format Location Object
5359 Session Initiation Protocol Service Examples
5373 Requesting Answering Modes for the Session Initiation Protocol
5379 Guidelines for Using the Privacy Mechanism for SIP
5806 Diversion Indication in SIP
5876 Updates to Asserted Identity in the Session Initiation Protocol (SIP)
5922 Domain Certificates in the Session Initiation Protocol (SIP)
6337 Session Initiation Protocol (SIP) Usage of the Offer/Answer Model
6432 Carrying Q.850 Codes in Reason Header Fields in SIP (Session Initiation Protocol) Responses
8224 Authenticated Identity Management in the Session Initiation Protocol (SIP)
8225 PASSporT: Personal Assertion Token
8226 Secure Telephone Identity Credentials: Certificates
8588 Personal Assertion Token (PaSSporT) Extension for Signature-based Handling of Asserted
information using toKENs (SHAKEN)

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 Transport protocols

Related links
IP Office SIP trunk specifications on page 875

Transport protocols
• UDP • RTP • RTCP
• TCP

Related links
IP Office SIP trunk specifications on page 875

Request methods
• INVITE • REFER • OPTIONS
• ACK • REGISTER • UPDATE
• BYE • SUBSCRIBE • PUBLISH
• CANCEL • NOTIFY • MESSAGE
• INFO • PRACK • PING

Related links
IP Office SIP trunk specifications on page 875

Response methods
• 100 Trying • 183 Session progress • 4XX
• 180 Ringing • 200 OK • 5XX
• 181 Call Is Being • 202 ACCEPTED • 6XX
Forwarded
• 3XX
• 182 Call Queued

Related links
IP Office SIP trunk specifications on page 875

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IP Office SIP trunk specifications

Headers
• Accept • Diversion • Proxy-Require
• Alert-Info • From • Require
• Allow • History-Info • Remote-Party-ID
• Allow-Event • Max-Forwards • Server
• Authorization • P-Asserted-Identity • Session-Timers
• Call-ID • P-Early-Media • Supported
• Contact • P-Preferred-Identity • To
• Content-Length • Privacy • User-Agent
• Content-Type • Proxy-Authenticate • Via
• CSeq • Proxy-Authorization • WWW-Authenticate

Additional Information
• The IP Office supports Call-ID headers of up to 256 characters.
• For IP Office R11.1 FP2 SP3 and higher, the maximum length of the tag element in From
and To headers has increased to 150 characters (previously 80 characters).
Related links
IP Office SIP trunk specifications on page 875

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Part 9: VMPro Auto Attendants

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Chapter 91: Voicemail Pro Auto-Attendants

From IP Office R11.1 FP2, the system supports auto-attendants provided by Voicemail Pro but
configured within IP Office Web Manager (these auto-attendants cannot be configured through IP
Office Manager).
• This is separate from the auto-attendant services supported on IP500 V2 systems using
embedded voicemail. Refer to the IP Office Embedded Voicemail Installation manual.
An auto-attendant consists of several greeting prompts that the callers hear and a set of definitions
of what the system should do when the caller presses any particular telephone key. Once you have
configured an auto-attendant, it can be used as the destination for incoming calls.
The system allows you to configure multiple auto-attendants:
• IP500 V2 systems support up to 40 auto-attendants.
• IP Office Server Edition and Select systems support up to 100 auto-attendants.
For each, you can configure which actions are performed when the caller presses a key 0 to 9, * and
#.
Feature Description
Greetings and Time Each auto-attendant can use time profiles to control which of up to 3 greeting is
Profiles played to a caller. This allows different greetings, such as “Good morning”, “Good
afternoon” or ““Sorry, we’re closed at the moment”” to be played based on the day
of the week, time of day or even specific dates.
The Menu Following the currently active greeting (if any), the caller hears the menu
Announcement announcement. This should list the auto-attendant actions that have been
configured. For example “Press 1 for .., press 2 for ....”.
Actions Separate actions can be defined for each of the standard telephone keys (0 to
9, * and #). Actions include transfer to a specified destination, transfer to another
auto-attendant, transfer to an extension specified by the caller, etc.
Text-to-Speech (TTS) For subscription mode systems, the greetings and menus used by the auto-
attendants can be generated using text-to-speech. This provides consistency in
the prompt voice used whilst be able to make rapid changes.
Automatic Speech For subscription mode systems, automatic speech recognition can be used to
Recognition (ASR) detect the caller's response to options provided by the auto-attendant.

Related links
Google TTS Prompt Language on page 881
Text-to-Speech (TTS) Prompts on page 881
Enabling Google Speech and the Default Voice on page 882
Auto-Attendant Fallback Options on page 883

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 Google TTS Prompt Language

Auto-Attendant Consent Example on page 883

Google TTS Prompt Language

Whilst the auto-attendant greeting and announcement prompts are recorded in a language of your
choice, some of the auto-attendant actions may play additional prompts provided by the system. In
that case, the language used for those system provided prompts is determined in several ways.
System Type Language Setting
Fixed Language If the system's Google Speech AI, or the auto-attendant’s/system conferences Speech
AI is set to a specific language, that language is used for all the system and TTS
prompts.
Call Locale Based If the system’s Google Speech AI language is set to Off, the language used for the
Language auto-attendant’s system prompts is determined from the locale associated with the call.
• Incoming Call Route Locale: If the caller is external, the incoming call route locale is
used if set.
• User Locale: If the caller is internal, the user locale is used if set.
• System Locale: If no user or incoming call route locale is set, the system locale is
used unless overridden by a short code locale.
• Short Code Locale: The short code locale is used, if set, if the call is routed to
voicemail using the short code. This overrides the other locale settings.

Related links
Voicemail Pro Auto-Attendants on page 880

Text-to-Speech (TTS) Prompts

Subscription mode systems can use text-to-speech (TTS) generated prompts for a range of
features. TTS is supported in a number of languages and with various voice choices, as shown in
the table below.
Whenever the settings for a TTS prompt are changed, the next time the prompt is requested,
including preview through the web administration menus, the resulting prompt is cached by the
system as a local file. This removes the initial delay that can occur the first time a TTS prompt is
played.
Language Number of Voices Genders
Arabic 3 Female and Male
Czech 1 Female only
Danish 1 Female only
Table continues…

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Voicemail Pro Auto-Attendants

Language Number of Voices Genders

Dutch 5 Female and Male
English (Australian) 4 Female and Male
English (United Kingdom) 4 Female and Male
English (United States) 6 Female and Male
Finnish 1 Female only
French (France) 4 Female and Male
French (Canada) 4 Female and Male
German 4 Female and Male
Greek 1 Female only
Hungarian 1 Female only
Italian 4 Female and Male
Japanese 4 Female and Male
Norwegian 5 Female and Male
Polish 5 Female and Male
Portuguese (Brazil) 1 Female only
Portuguese (Portugal) 4 Female and Male
Spanish 1 Female only
Swedish 1 Female only
Turkish 5 Female and Male

Related links
Voicemail Pro Auto-Attendants on page 880

Enabling Google Speech and the Default Voice

About this task
Subscription systems can use Google speech to provide text-to-speech prompts and automatic
speech recognition. These can be used with auto-attendants and system conferences.
Note:
• When enabled, the Google TTS is used for all Voicemail Pro TTS functions, overriding
any locally installed TTS service.
Procedure
1. Select System > Voicemail.
2. Enable Google Speech AI.

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 Auto-Attendant Fallback Options

3. Select the default Speech Language and Speech Voice the system should use.
• The choices are used as the system defaults. They can be overridden within each
auto-attendant. The language can be overridden within Voicemail Pro call flows.
4. Save the updated settings.
Related links
Voicemail Pro Auto-Attendants on page 880

Auto-Attendant Fallback Options

Whilst auto-attendants are intended to let callers choose the required destination for their call
themselves, there may be cases where this fails. For example, when the system does not detect
any response from the caller or when it cannot match the response to any of its configured
options.
There are a number of fallback routes that can be applied to calls in such scenarios:
Stage Fallback Route
Fallback Action This option is used when the number of times the auto-attendant has waited for a
valid response exceeds the Menu Loop Count. It can be configured to perform a
chosen auto-attendant action. Otherwise, the system ends the call.
• Note that is option may be overridden by the Maximum Inactivity timeout if it is
reached first. See ‘External Call Fallback’ below.
Park and Page Fallback If the caller selects a Park & Page action, their call is parked and waits to be
Number unparked. If the call is still parked after a number of page attempts configured for
the action, it is transferred to the action's configured Fallback Number.
External Call Fallback This is an incoming call route setting. For external calls routed to an auto-attendant
Extension from an incoming call route, it is used if for some reason the auto-attendant service
is not available.

Related links
Voicemail Pro Auto-Attendants on page 880

Auto-Attendant Consent Example

In the following example, the business wants to record external calls to its sales group using the
group’s automatic recording settings. However, it needs to provide those callers with an option to
opt-out of being recorded and to have that choice recorded in the system’s log files.
Two sales groups are configured. Each with the same members but only one with automatic call
recording of external calls configured.
A Sales auto-attendant was added.

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Voicemail Pro Auto-Attendants

Within the auto-attendant, the menu announcement prompt informs the callers of the option to not
be recorded.

The auto-attendant’s actions then route the caller either to the group that has recording enabled or
the group that does not support recording. The consent settings of the actions record the caller’s
choice in the system’s log files.

Related links
Voicemail Pro Auto-Attendants on page 880

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Chapter 92: Voicemail Pro Auto-Attendant
Settings

Call Management > Auto Attendants > /+Add

This section describes the auto–attendant settings used for subscription systems using Voicemail
Pro. For auto-attendants provided by embedded voicemail on IP500 V2 systems, see Auto
Attendant section.
These are split into two tabs.
Tab Description
Auto Attendants This tab defines the general settings of the auto-attendant and its greetings and
announcements.
Action This tab defines the functions provided by the individual telephone keys.

Related links
Auto-Attendant on page 885
Actions on page 889

Auto-Attendant
These settings are used to define the operation of the auto–attendant service whilst it waits for the
caller to select an option from the configured actions.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For IP Office R11.1 FP2 and higher, auto-attendants are also supported on systems that use
Voicemail Pro. However, the configuration of those auto-attendants is done using IP Office Web
Manager.

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Voicemail Pro Auto-Attendant Settings

Auto-Attendant Settings
Field Description
Name Range = Up to 12 characters
The name for the auto-attendant. Set a name that acts as a reminder of the auto-
attendants role. The name is then also shown in other menus used to route calls to the
auto-attendant.
AA Number This number is automatically assigned by the system and cannot be changed. It is used in
conjunction with short codes to access the auto–attendant service or to record greetings.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
• IP500 V2 systems support up to 40 auto-attendants.
• IP Office Server Edition and Select systems support up to 100 auto-attendants.
Maximum Default = 8 seconds; Range = 1 to 20 seconds.
Inactivity
This value sets how long the attendant should wait for a response from the caller after
playing any current prompts.
• If the caller responds, their response is checked for a match to a configured action
without any further wait.
• Note that the caller can respond whilst the prompts are playing.
• If the timeout expires, the Menu Loop Count is checked to determine the next steps.
Name Match Default = Last then First
Order
This setting sets the name order used for the Dial By Name action if used.
Direct By Number Default = No
This setting affects the operation keys set to the Dial By Number action.
• If enabled: The caller’s key press to select the action is included in the digits they dial
for a extension match. For example, if menu key 2 is used for the action, a caller can
dial 2 and then 01 for extension 201.
• If not enabled: The caller’s key press to select the action is not included in the digits
they dial for extension match. For example, if menu key 2 is used for the action, a caller
must dial 2 and then 201 for extension 201.
Direct By Default = No
Conference
This setting affects the operation keys set to the Dial By Conference action.
• If enabled: The caller’s key press to select the action is included in the digits they dial
for a conference match. For example, if menu key 3 is used for the action, a caller can
dial 3 and then 01 for conference 301.
• If not enabled: The caller’s key press to select the action is not included in the digits
they dial for a conference match. For example, if menu key 3is used for the action, a
caller must dial 3 and then 301 for conference 301.
Table continues…

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 Auto-Attendant

Field Description
Enable Local Default = Yes
Recording
When off, use of short codes to record auto-attendant prompts is blocked. The short
codes can still be used to playback the greetings.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
Speech AI Default = Off
This option is only available on subscription mode systems. It sets whether the auto-
attendant supports text-to-speech and automatic speech recognition features.
• When off, the auto-attendant does not support any text-to-speech and speech
recognition features.
- The language used for any prompts provided by the system is determined from the
call settings. See Google TTS Prompt Language on page 881.
• When set to a specific language, the auto-attendant supports text-to-speech and speech
recognition features in that language.
- It also uses that language for all system prompts it provides regardless of the locale
call settings the system has associated with the call.
Speech Voice This setting is available when Speech AI is set to a specific language. It allows selection
of a particular voice used for any text-to-speech features.
See Text-to-Speech (TTS) Prompts on page 881.

Greeting and Announcement Settings

When a caller reaches an auto-attendant, they first hear the attendant's current greeting (if any)
and then the attendant's menu announcement.
• The greeting used is the first one (from up to 3 defined greetings) for which the greeting's
associated time profile is currently active. This allows you to define greetings for different
times of day (for example “Good Morning”, “Good Afternoon” and “Sorry, we are currently
closed”) or different greetings for business and non-business days.
• The menu announcement should contain the instructions for the caller regarding the keys
they can press and other actions.
• Each time a caller goes round the auto-attendant loop, they can respond (with key presses or
speech) whilst any greeting and announcement menu prompt is being played.
Field Description
Optional Greeting Up to 3 greetings can be defined using the Add Greeting button.
1
• Each greeting requires an associated time profile.
Optional Greeting
2 - Time Profile: Default = Off (Greeting not used).
• If Off, the greeting is not used.
• The greeting is only used when defined by its associated time profile.
• When multiple greetings are defined, the first one that has an active time profile, in the
order 1 to 3, is used as the current greeting.
Table continues…

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Voicemail Pro Auto-Attendant Settings

Field Description
Optional Greeting • If no greetings is currently active according to its time profile, then no greeting is played.
3 • If a greeting is no longer required, it can be deleted by clicking on the adjacent icon.
• After playing any greeting, the system always then plays the menu announcement.
Menu The menu announcement should contain the instructions for callers about the actions they
Announcement can perform. For example; “Press 1 for reception. Press 2 for sales, ...””
It is used as follows:
• When a call first reaches the auto-attendant, it is played to the caller after whichever
greeting is currently active.
• If the Menu Loop Count is not zero, it is played again at the start of each repeat loop.
• The caller can respond by pressing a key whilst the announcement is being played. On
subscription mode systems, if Speech AI is enabled they can also respond by speaking
whilst the announcement is played.
• After the announcement is played, the auto-attendant waits for a response for the time
set by the Maximum Inactivity setting.
Menu Loop Count Default = 0 (No Repeat)
This setting sets the number of times the auto-attendant will repeat the Menu
Announcement and then wait for a valid response.
If the caller does not respond or their response is not matched to an action:
• If 0, the default, they hear the No Match Prompt prompt and the Fallback Action
setting is used.
• If non-zero but the number of repeat loops has not been reached, they hear the No
Match Prompt and then the Menu Announcement again and the auto-attendant waits
for a response again.
• If non-zero and the number of repeat loops has been reached, they hear the No Match
Prompt prompt and the Fallback Action setting is used.
No Match Prompt This prompt is heard when the caller does not respond in time or if their response does
not match a configured action. For example; “Sorry, no response was recognized.”
• Note that this prompt is also heard by callers who are about to be redirected to the
Fallback Action. Therefore a prompt like “"Please try again"” would not be appropriate.

The following settings are common to the menu announcement, greetings and error message.
The greetings and announcements can be recorded from the phone, use an uploaded file or be
provided by text-to-speech. Whichever method was last used or configured overrides any previous
prompt.
Field Description
Dial To Record Default = Automatically assigned. Not changeable.
Greeting
This field indicates the short code that can be dialed in order to record the greeting from
an internal extension.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
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 Actions

Field Description
Audio Output Default = Audio File
The field sets the current method used to provide the prompt used for the greeting or
announcement. Clicking on the current value allows you to see its current settings and to
change them or to change the recording method.
• Audio File (wav) – Provide the prompt using a pre-recorded audio file.
See Using Pre-Recorded Prompt Files on page 907.

Note:
Use IP Office Web Manager to upload the .wav file.
• Text To Speech – Provide the prompt using the text-to-speech service. This option
is only available on subscription mode systems with Speech AI enabled and set to a
specific language.
See Recording Auto-Attendant Prompts Using Text-to-Speech on page 908.

Related links
Voicemail Pro Auto-Attendant Settings on page 885

Actions
This tab defines the actions available to callers dependent on which DTMF key they press or, on
subscription mode systems, based on automatic speech recognition of keywords. To change an
action, click on the appropriate button.
The Fallback Action action applied is the user does not make a recognized choice is configured
separately through the No Match Prompt prompt settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Settings: Keys/Events
The following actions can be assigned to the selected keys.
Action Description
0 to 9, *, # These keys correspond to the standard telephone dial pad key. Clicking on a key allows
configuration of its settings.
Fax If configured, the Fax option is used when the system detects fax tone.
Table continues…

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Voicemail Pro Auto-Attendant Settings

Action Description
Fallback Action Default = Drop Call
This option is used when the number of times the auto-attendant has waited for a valid
response from the caller has exceeded the Menu Loop Count. It is preceded by the No
Match Prompt and then the configured action is performed.
All actions are supported except Park & Page, Replay Menu Greeting, Speak By Name
and Speak By Number
You can choose whether to mention this option in the Menu Announcement. For
example, if set to transfer to your receptionist, add “... or wait to for our operator.”
Menu The menu announcement should contain the instructions for callers about the actions they
Announcement can perform. For example; “Press 1 for reception. Press 2 for sales, ...””
It is used as follows:
• When a call first reaches the auto-attendant, it is played to the caller after whichever
greeting is currently active.
• If the Menu Loop Count is not zero, it is played again at the start of each repeat loop.
• The caller can respond by pressing a key whilst the announcement is being played. On
subscription mode systems, if Speech AI is enabled they can also respond by speaking
whilst the announcement is played.
• After the announcement is played, the auto-attendant waits for a response for the time
set by the Maximum Inactivity setting.

Settings: Key Actions

Action Description
Not configured Perform no action.
Dial By Allow the caller to dial the conference ID they require.
Conference
See Dial By Conference on page 892.
Dial By Name Prompt the caller to dial the name of the user they require.
See Dial By Name on page 893.
Dial By Number Allow the caller to dial the extension number they require.
See Dial By Number on page 895.
Leave Message Redirect the caller a specified mailbox to leave a message.
See Leave Message on page 896.
Supervised Transfer the caller to the specified extension number.
Transfer
See Supervised Transfer on page 897.
Park & Page Park the call and make an announcement to the a specified group.
See Park & Page on page 898.
Table continues…

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 Actions

Action Description
Replay Menu Replay the auto-attendant’s menu announcement.
Greeting
See Replay Menu on page 900.
Unsupervised Transfers the caller to the specified extension number.
Transfer
See Unsupervised Transfer on page 903.
Transfer To Auto Transfers the caller to another auto–attendant.
Attendant
See Transfer to Auto Attendant on page 904.
Speak By Name Allow the caller to select from listed names using speech.
See Speak By Name on page 901.
Speak By Number Allow the caller to speak the extension number required.
See Speak By Number on page 902.
Destination The destination depends on the action:
• Leave Message, Supervised Transfer and Unsupervised Transfer – Use the drop-
down to select the target extension.
• Transfer To Auto Attendant – Use the drop-down to select another existing auto-
attendant.
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Settings on page 885

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Chapter 93: Voicemail Pro Auto-Attendant
Actions

The following sections provide more details on the different auto-attendant actions that can be
assigned to the keys 0 to 9, # and *.
Related links
Dial By Conference on page 892
Dial By Name on page 893
Dial By Number on page 895
Leave Message on page 896
Supervised Transfer on page 897
Park & Page on page 898
Replay Menu on page 900
Speak By Name on page 901
Speak By Number on page 902
Unsupervised Transfer on page 903
Transfer to Auto Attendant on page 904

Dial By Conference
This action allows the caller to select the conference they want to join by dialing the conference
ID. For example, “If you know the conference you want, dial the conference number.”.
The behavior of the action depends on the auto-attendant’s Direct By Conference setting.
• If enabled: The caller’s key press to select the action is included in the digits they dial for a
conference match. For example, if menu key 3 is used for the action, a caller can dial 3 and
then 01 for conference 301.
• If not enabled: The caller’s key press to select the action is not included in the digits they
dial for a conference match. For example, if menu key 3is used for the action, a caller must
dial 3 and then 301 for conference 301.

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 Dial By Name

Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Dial By Name
This action allows callers to dial the name that want and then hear a list of matches from which
they can make a selection. For example, “To select from a list of names, press 1”.
Callers selecting this option are asked to dial the name of the user they require and then press #.
They then hear a list of possible matches from which they can make a selection. The list uses the
recording mailbox name prompts of matched users.
• The name matching uses the auto-attendant’s Name Match Order setting to either match
against first or last names.
• The name used for matching is the user’s Full Name if set, otherwise their Name is used.
Users are excluded from matching if they:
• Are marked as Ex Directory in their user settings.
• Do not have a recorded mailbox name prompt. Normally user’s are asked to record a name
when they first access their mailbox. See Recording User Name Prompts on page 908.

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Voicemail Pro Auto-Attendant Actions

Dial By Name assumes that a standard ITU lettered dialing pad is being used.
ABC D EF

1 2 3
GHI JKL MNO

4 5 6
PQRS TUV WXYZ

7 8 9

* 0 #

How Dial By Name works

1. The caller is prompted to dial the name they require and then press #.
• For example: Dialing 527 matches names starting with JAS (for example "Jason") and
KAR (for example "Karl").
• Callers can also press *# to exit without making a selection.
2. Depending on the number of matches found:
• If no matches are found, the caller is given the option to retry.
• If 10 or less matches are found, the matching mailbox name greetings are played as
part of a list. For example, “”Press 1 for …, Press 2 for …, Press 3 for …””.
• If more then 10 matches are found, the caller is prompted to either press # to hear the
first 10 matches or to dial more characters to reduce the number of matches. If they
select to play the list, after each set of 10 matches they can either make a selection or
follow the prompts for other options.
Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
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 Dial By Number

Key Description
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Dial By Number
This action allows the caller to select the extension they want by dialing the extension number. It
can be used to allow callers to directly access user and group extension numbers.
For example, “If you know the extension you want, dial the extension number.” or “If you know the
extension you want, press 1 followed by the extension number”.
The behavior of the action depends on the auto-attendant’s Direct By Number setting.
• If enabled: The caller’s key press to select the action is included in the digits they dial for a
extension match. For example, if menu key 2 is used for the action, a caller can dial 2 and
then 01 for extension 201.
• If not enabled: The caller’s key press to select the action is not included in the digits they
dial for extension match. For example, if menu key 2 is used for the action, a caller must dial
2 and then 201 for extension 201.

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Voicemail Pro Auto-Attendant Actions

Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Leave Message
This action directs the caller to the mailbox of the specified extension (user or group). For
example, “To leave a message, press 1”.
The caller hears the mailbox’s prompt and is then asked to leave a message.
Action Settings
Key Description
Destination The selected destination for the mailbox into which the message should be left. The
feature can be used to leave messages in mailboxes where the user/group does not have
Voicemail On enabled.
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 Supervised Transfer

Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Supervised Transfer
This action transfers the caller to the specified extension number (user or group). Once
transferred, the caller is handled the same as a normal call to the same number. For example;
queuing, following any forwards, etc.
Action Settings
Key Description
Destination The selected destination for the transfer. This action can be used with or without a set
destination:
• When no destination is set, the action behaves like Dial By Number above.
• When a destination is set, the action waits for a connection before transferring the call.
• Whilst waiting, the caller hears the system’s music on hold.
Table continues…

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Voicemail Pro Auto-Attendant Actions

Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Park & Page

This action parks the caller’ whilst the system performs a page to a specified user or group
extension number. The page message includes the park slot number assigned to the parked call
so that anyone who hears the page can unpark it.
• Whilst parked, the caller hears music on hold.
• The system uses the prompt you have configured for the button to announce that there is a
parked call. It then states the park slot number which can be used to unpark the call.

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 Park & Page

Action Settings
Key Description
Park Slot Prefix The park slot prefix number. Maximum is 8 digits. A 0-9 will be added to this prefix to form
a complete park slot ID for the parked call.
The system the park slot prefix to create park slot for a call by adding an extra digit (0-9).
For example, if you set 62080 as the prefix, the system uses a number between 620800
and 620809 to park calls.
Paging Number Select the user or group which the system will page in order to announce the parked
caller.
Retry Count The number of page retries. The range is 0 to 5.
Retry Timeout Default = 15 seconds.
The time, in minutes and seconds, between paging retires. The value can be set in 15
second increments up to a maximum of 5 minutes. The default is 15 seconds.
Fallback Number The extension number to which the park called should be presented if, after the final page
and retry timeout, the call is still parked.
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

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Voicemail Pro Auto-Attendant Actions

Field Description
Dial To Record Default = Automatically assigned. Not changeable.
Greeting
This field indicates the short code that can be dialed in order to record the greeting from
an internal extension.
See Recording Auto-Attendant Prompts Using Short Codes on page 906.
Audio Output Default = Audio File
The field sets the current method used to provide the prompt used for the greeting or
announcement. Clicking on the current value allows you to see its current settings and to
change them or to change the recording method.
• Audio File (wav) – Provide the prompt using a pre-recorded audio file.
See Using Pre-Recorded Prompt Files on page 907.

Note:
Use IP Office Web Manager to upload the .wav file.
• Text To Speech – Provide the prompt using the text-to-speech service. This option
is only available on subscription mode systems with Speech AI enabled and set to a
specific language.
See Recording Auto-Attendant Prompts Using Text-to-Speech on page 908.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Replay Menu
This action replays the auto-attendants Menu Announcement recording. For example, “To hear
the options again, press #”.
Replaying the greeting does not count as a loop for auto-attendant’s Menu Loop Count.

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 Speak By Name

Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Speak By Name
This action is only available on subscription systems and when Speech AI is set to a specific
language (enabling support for speech recognition).
This action is similar to Dial By Name. However, when the caller is presented with a list of name
matches, they can indicate their selection by speaking.

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Voicemail Pro Auto-Attendant Actions

Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Speak By Number
This action is only available on subscription systems and when Speech AI is set to a specific
language (enabling support for speech recognition).
This action is similarly to Dial By Number. However, the caller can dial or speak the extension
number that they require. Note that it does not use the Direct By Number setting.

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 Unsupervised Transfer

Action Settings
Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Unsupervised Transfer
This action transfers the caller to the specified extension number (user or group). Once
transferred, the caller is handled the same as a normal call to the same number. For example;
queuing, following any forwards, etc.
Action Settings
Key Description
Destination The selected destination for the transfer. Unlike the Supervised Transfer action, this
action cannot be configured without a destination.
Table continues…

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Voicemail Pro Auto-Attendant Actions

Key Description
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

Transfer to Auto Attendant

This action transfers the caller to another auto-attendant. For example, ““For alternate options,
press #””.
Up to 40 auto-attendants can be configured and linked.
Action Settings
Key Description
Destination The selected auto-attendant.
Speech This option is only available on subscription mode systems and when Speech AI is set to
Recognition a specific language. It allows the action to be triggered by speech recognition of keywords.
Keywords
• The keywords must be unique. The same word cannot be used for another key.
• Up to 3 keywords are supported per key, separated by commas. Note that using more
keywords in total reduces the chances of a match.
• Avoid using proper names. These are less likely to be matched as they may not match
existing words in the speech recognition dictionaries used by Google.
• Encourage matches by ensuring that the keywords are part of the announcements
played to callers. For example, “Say whether you want Sales or Support” rather than
“Say what department you want”.
Table continues…

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 Transfer to Auto Attendant

Key Description
Consent Directive When a caller selects a particular action, the actions Consent Directive value is included
in the system logs. These options allow you record whether the caller has indicated their
consent to some action, for example call recording.
See Auto-Attendant Consent Example on page 883.
• Consent Not Applicable – Indicates that the caller has not been prompted to select
whether they consent to call recording.
• Consent Given – Indicates that the caller has been prompted for their consent and has
consented.
• Consent Denied – Indicates that the caller has been prompted for their consent and
has not consented.

Related links
Voicemail Pro Auto-Attendant Actions on page 892

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Chapter 94: Recording Auto-Attendant
Prompts (Voicemail Pro)

The prompts used by the auto-attendant can be provided through a number of methods.
Related links
Recording Auto-Attendant Prompts Using Short Codes on page 906
Using Pre-Recorded Prompt Files on page 907
Recording Auto-Attendant Prompts Using Text-to-Speech on page 908
Recording User Name Prompts on page 908

Recording Auto-Attendant Prompts Using Short Codes

The Dial To Record Greeting values shown in the auto-attendant menus indicates a short code
which can be used to play and record the associated auto-attendant prompt.
• These short codes can be dialed from any internal extension.
• The short codes can only be used if the auto-attendants Enable Local Recording setting is
enabled.
• Recording a prompt using this method overrides any previously uploaded audio file or TTS
settings for the prompt.
Using a Short Code
When using the short codes, you will be prompted as follows:
• Press 1 to hear the current recorded prompt if any.
• Press 2 to record a new prompt.
- After the tone, record the prompt. Note that the prompt must be at least 3 seconds in
length.
- Press 2 again to end recording.
• Press 3 to save the new prompt.
Short Codes List

• Optional Greeting 1 – Dial *81 followed by the AA Number . For example, *8101 for the first
auto-attendant.

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 Using Pre-Recorded Prompt Files

• Optional Greeting 2 – Dial *82 followed by the AA Number. For example, *8201.
• Optional Greeting 3 – Dial *83 followed by the AA Number. For example, *8301.
• Menu Announcement – Dial *84 followed by the AA Number. For example, *8401.
• No Match Prompt – Dial *87 followed by the AA Number. For example, *8701.

• Park & Page Prompts – Dial *80 followed by the action key being used (0 to 9) and then the
AA Number. For example, for a park & page action on button 2 of the first auto-attendant,
dial *80201. These prompts are used as part of the page call made by the system.
- For the * key, dial *8510 followed by the AA Number. For example, *851001 for the first
auto-attendant.
- For the # key, dial *8511 followed by the AA Number. For example, *851101.
How are the Dialing Codes Configured?
The dialing codes use system short codes which are automatically added to the system
configuration when the first auto-attendant is created. Editing or deleting those system short codes
will affect the operation of the codes shown in the auto-attendant menus.
These short codes use the Auto Attendant feature.
Related links
Recording Auto-Attendant Prompts (Voicemail Pro) on page 906

Using Pre-Recorded Prompt Files

You can use pre-recorded audio files as prompts for the auto-attendant.
• Prompt file uploading is only supported when using IP Office Web Manager. It cannot be
done from the IP Office Manager menu.
• The file must be a .wav file in Mono PCM 16-bit format, either 8, 16 or 22KHz. Maximum
length 10 minutes.
To upload an audio file:
1. Note that uploading a file will override any previously recorded audio file or TTS setting.
2. For the greeting and menu announcement prompts, click on the Audio Output and select
Audio File (wav).
3. Click on Upload and select the recording file. Alternatively, drag and drop the file into the
text box.
4. Click on Upload.
5. Use the playback controls to test the recording.
Related links
Recording Auto-Attendant Prompts (Voicemail Pro) on page 906

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Recording Auto-Attendant Prompts (Voicemail Pro)

Recording Auto-Attendant Prompts Using Text-to-Speech

On subscription mode systems, Text-to-Speech (TTS) can be used to provide the auto-attendant
greetings and menu announcement prompts.
• TTS prompts are only available when Google Speech AI is enabled. See Enabling Google
Speech and the Default Voice on page 882.
• The language used for TTS prompts is set by the auto-attendant's Speech AI setting.
• The voice used is set by the auto-attendant's Speech Voice setting. See Text-to-Speech
(TTS) Prompts on page 881.
• You can enter up to 250 words as a prompt.
• Commas are treated as a short pause, semi-colons as a long pause.
• Following any changes, once a prompt is played or previewed, it is cached by the system in
order to remove any future playback delay.
To create a TTS prompt:
1. Note that configuring TTS will override any previously recorded audio file.
2. Click on the Audio Output and select Text To Speech.
3. Enter the required text into the text box.
• Use a comma to add a short pause.
• Use a period to add a long pause.
• To add emphasis to a particular word, add _ underscores before and after the word.
4. Make any changes required to the text.
Related links
Recording Auto-Attendant Prompts (Voicemail Pro) on page 906

Recording User Name Prompts

The Dial By Name and Speak By Name features only include users who have recorded a
mailbox name (and are not set as ex-directory). By default, users are asked to record a name
when they first access their mailbox.
However, in some scenarios this may need to be done separately. The method for recording the
user name depend on how they access their mailbox and the mode in which the voicemail service
is running.
Visual Voice
If the user accesses voicemail mailbox using the visual voice menu on their phone, they can use
the following process to record their name:
1. Access visual voice.
2. Scroll down to and select Name.

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 Recording User Name Prompts

3. Record a name.
4. When happy with the recording, press Select.
Intuity Mailbox Mode
If the user access their voicemail mailbox using spoken prompts, for example by dialing *17, they
can use the following process to record their name:
1. Access the mailbox prompts.
2. Press 5.
3. Press 5 again.
4. The user will hear their current name recording, if any.
5. After the tone, record a name and press 1.
6. The name is played again.
• To accept the recording, press #.
• To record the name again, press 1.
IP Office Mailbox Mode
If the user access their voicemail mailbox using spoken prompts, for example by dialing *17, they
can use the following process to record their name:
1. Access the mailbox prompts.
2. Press *05 to select the option to record your name.
3. Press 1 to hear your current recording.
4. Press 2 to record your name. When prompted, speak your name. The maximum recorded
length is 5 seconds.
5. Press 2 when you have finished recording your name.
6. Press 1 to listen to your new recording. Review the recording and select one of the
following options:
• To save the new recording: Press 3.
• To record your name again: Press 2.
Related links
Recording Auto-Attendant Prompts (Voicemail Pro) on page 906

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Chapter 95: Routing Calls to a Voicemail
Pro Auto-Attendant

This section provides notes on the different methods by which calls can be directed to a Voicemail
Pro auto-attendant.
Related links
Routing External Calls to an Auto-Attendant on page 910
Routing Internal Calls to an Auto-Attendant on page 910

Routing External Calls to an Auto-Attendant

Once an auto-attendant has been created, it becomes selectable as a destination in other menus,
for example incoming call routes. This is shown by entries prefixed with AA: in the drop-down lists
of selectable destinations.
Related links
Routing Calls to a Voicemail Pro Auto-Attendant on page 910

Routing Internal Calls to an Auto-Attendant

Typically auto-attendants are not used for handling internal calls. However, it can be useful:
• To test the operation of an auto-attendant whilst it is being configured.
• As a number to which users can transfer external caller's who have been misdirected.
Short Codes for Auto-Attendant Access
An internally dialable number to access an auto-attendant can be created using the Auto
Attendants short code feature. For example:
• *99XX/Auto Attendant/"AA:"N - This short code will allows calls to any auto-attendant using
the AA Number when dialing. For example, *9901 for the first auto-attendant.
• *99/Auto Attendant/"AA:AutoAttend01" - This short code allows calls to a specific auto-
attendant using the auto-attendant Name setting.

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 Routing Internal Calls to an Auto-Attendant

Related links
Routing Calls to a Voicemail Pro Auto-Attendant on page 910

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Part 10: Conferencing

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Chapter 96: Conferencing

The system supports a range of conference call features.

Related links
Conference Types on page 913
Conference Participants on page 914
User Conference Controls on page 914
Conference Capacities on page 915
Conference ID Numbers on page 916
Conference Notes on page 916
Conference Phones on page 917
Context Sensitive Conferencing on page 918

Conference Types
The system supports conferences consisting of multiple internal and external parties.
Conference Type Description
Ad-Hoc An ad-hoc conference is one created by the system on the fly. For example, when a
Conferences user with two calls in progress conferences those calls using their phone. For ad-hoc
conferences, all internal users are treated as moderators.
See Ad-Hoc Conferencing on page 921.
Meet–Me A meet-me conference is one started using a specific fixed conference ID number.
Conferences This allows the use of various features to route and place calls into specific meet-me
conferences.
Personal Meet-Me Each user's own extension number is treated as their personal meet-me conference
Conference number. That user is the conference’s only moderator. Other participants can join a
personal meet-me conference at any time, however the audio conference only starts
when the owner also joins. If the user’s optional conference PIN has been configured,
the system prompts other callers for the PIN when they try to access the personal
meet-me conference.
See Personal Meet-Me Conferences on page 923.
Table continues…

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Conferencing

Conference Type Description

System Meet-Me System meet-me conferences are configured by system administrators. Each system
Conferences conference has a fixed conference ID and appears in the list of available destinations for
auto-attendant actions, DDI numbers, incoming call routes, etc. Each system conference
can be configured with multiple moderators, separate PINs for moderators and other
participants, etc.
See System Conferences on page 927.

Related links
Conferencing on page 913

Conference Participants
The following terms are used for the different roles people can have within a conference.
• Participant – Any member of a conference.
• Delegate – Any participant of a conference who is not a moderator.
• Moderator – Moderators have extra functions. For example they can drop and mute other
participants. Who is or can be a moderator depends on the conference type:
- Ad-Hoc Conferences – Any internal participant is automatically also a moderator.
- Personal Meet-Me Conferences – The conference owner is the only moderator.
- System Conferences – A participant of a system conference can be become a moderator
is either of 2 ways:
• Specified internal users can be added to the conference’s moderator list. Those users
are automatically moderators.
• If the optional moderator PIN is set, any caller who enters that PIN joins the conference
as a moderator. This allows external callers to be moderators (though without ability to
drop/mute other participants).
• Owner – Personal meet-me conferences are owned by the user with the same extension
number as the conference ID. They are also automatically the conference’s only moderator.
Related links
Conferencing on page 913

User Conference Controls

Internal users who join a conference may also have access to controls which allow them to
mute/unmute other parties and to drop other parties. The range of controls will depend on the
conference type and whether the user is a moderator or a delegate.

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 Conference Capacities

Phone Controls
Users with Avaya 1400, 1600, 9500, 9600 Series and J100 Series phones (except the J129) can
view the list of conference participants. Using the list, they can access options to mute and drop
themselves and other participants.
On these phones, programming Conference Meet Me buttons allows the user to receive
indication of when a particular conference is in progress and to access that conference.
User Portal Controls
Users with access to the User Portal can display details of the access settings for their own
personal meet-me conference and for any system conferences for which they have been added to
the moderator list. They also receive notification when other participants have joined their personal
meet-me conference and are waiting for them to join.
When they join any conference, the portal displays a list of participants and controls for muting/
dropping participants.
one-X Portal
This application provides the user with a display of conference participant and controls to manage
their conference participation. It can also provide the user with controls for scheduling conferences
and sending invitations to other conference participants.
SoftConsole
This application display details of conferences in progress to assist with transferring callers into a
conference. It also provides menus for starting two meet-me conferences.
Related links
Conferencing on page 913

Conference Capacities
For full details on system capacities, refer to Avaya IP Office™ Platform Guidelines: Capacity.
The following table summarizes the overall system capacity for conference calls and maximum
participants in any individual conference call. This capacity limits apply to all conference types.
System Mode Total Conference Participants Maximum Conference Size
IP Office Server Edition 256 256
IP Office Select 512 256
IP Office Subscription
IP500 V2 128 64

System Meet-Me Conferences

System meet-me conferences use the same resources as above. However, in addition there are
limits on the number that can be configured.

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Conferencing

Maximum Configured
IP500 V2 30
Other networks 120

In an IP Office Server Edition/Select network, these conferences are hosted on the primary server.
If a secondary server is present, that server will host the system conferences during primary
server resilience.
Related links
Conferencing on page 913

Conference ID Numbers
Every conference is assigned a conference ID number. That number can be used with other
features (short codes, programmable buttons) in order to join that conference.
• Ad-hoc conferences are automatically assigned a conference ID number when started. Each
ad-hoc conference uses the first available ID from 100 upwards.
• Meet-me conferences use pre-set conference IDs set as follows:
- Personal meet-me conferences use a conference ID that matches the extension number of
the conference owner and moderator.
- System meet-me conferences use the conference ID specified when the conference
settings are configured.
• It is advisable not to use conference ID’s that are near the range that may be in use for
ad-hoc conferences as above (100 plus). Once a conference ID is in use by an ad-hoc
conference, it is no longer possible to join the conference using the various conference
meet me features.
Related links
Conferencing on page 913

Conference Notes
Feature Details
Other Uses System features such as call intrusion, call recording and silent monitoring all use
of Conference conference resources for their operation. On IP500 V2 systems, each Embedded
Resources Voicemail call in progress also reduces the conference capacity.
Table continues…

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 Conference Phones

Feature Details
Automatically The behavior for the system automatically ending a conference varies as follows:
Ending
• A conference remains active until the last extension or trunk with reliable disconnect
Conferences
leaves. Connections to voicemail or a trunk without reliable disconnect (for example an
analog loop-start trunk) will not hold a conference open.
• The Drop External Only Impromptu Conference setting controls whether a
conference is automatically ended when the last internal party exits the conference.
Analog Trunk In conferences that include external calls, only a maximum of two analog trunk calls are
Restriction supported. This limit is not enforced by the system software.
Recording If call recording is supported, conference calls can be recorded just like normal calls.
Conferences Note however that recording is automatically stopped when a new party joins the
conference and must be restarted manually. This is to stop parties being added to a
conference after any "advice of recording" message has been played.
IP Trunks and Conferencing is performed by services on the system's non-IP interface. Therefore a
Extensions voice compression channel is required for each IP trunk or extension involved in the
conference.
Call Routing A short code routing calls into a conference can be used as an Incoming Call Route
destination.
Conference Tones The system provides conference tones. These will be either played when a party
enters/leaves the conference or as a regularly repeated tone. This is controlled by the
Conferencing Tone (System | Telephony | Tones & Music) option.

Related links
Conferencing on page 913

Conference Phones
The system does not restrict the type of phone that can be included in a conference call.
Feature Details
Use Mute When not speaking, use of the mute function helps prevent background noise from
your location being added to the conference call. This is especially important if you are
attempting to participate handsfree.
Handsfree While many Avaya telephones can be used fully handsfree during a call, that mode
Participation of operation is intended only for a single user, seated directly in front of the phone.
Attempting to use a handsfree phones for multiple people to listen to and participate in a
call will rarely yield good results. See below for details of conference phones supported
by the system.
Table continues…

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Conferencing

Feature Details
Dedicated To allow multiple people in one room to speak and listen to a conference call, the system
Conference supports the following conference phones:
Phones
• B100 Conference Phones (B179 and B199).
• Audio Conferencing Unit (ACU).
Group Listen The Group Listen function can be used via a programmable button or short code. It
allows the caller to be heard through a phones handsfree speaker while only being talked
to via the phone's handset.

Related links
Conferencing on page 913

Context Sensitive Conferencing

On 1400, 1600, 9500, 9600 and J100 Series telephones there have been changes to the display
and handling of calls put on hold pending transfer. For those phones there have also been
changes to which calls are conferenced when a Conference button or Conf display option is
pressed on the telephone.
• Previously, pressing Conference would put the user's current call and all held calls into a
conference. That included any calls that had be put on hold pending transfer by pressing
Transfer.
• The result of pressing Conference on the telephone now depends on which call is currently
highlighted on the phone display and what other calls are held or held pending transfer.
Which call is Other condition (in Result when Calls Conferenced
highlighted on the priority order) Conference is Connected Held Calls Held
display pressed: Call Pending
Transfer
Connected call No call held Conferences the –
pending transfer connected call and
all held calls.
Call held pending Conferences the –
transfer connected call and
the held pending
transfer call. Any
other held calls are
unaffected.
Table continues…

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 Context Sensitive Conferencing

Which call is Other condition (in Result when Calls Conferenced

highlighted on the priority order) Conference is Connected Held Calls Held
display pressed: Call Pending
Transfer
Held call Connected call Conferences the – –
held call and
the connected call.
Any other held
calls including held
pending transfer are
unaffected.
Held pending Conferences the – –
transfer call held and held
pending transfer
call. All other held
calls are unaffected.
Held calls Conferences with all – –
other held calls.
Held pending Connected call Conferences the –
transfer call held pending
transfer call to a
connected call. Any
other held calls are
unaffected.
Held calls Conferences the –
call held pending
transfer with all
other held calls.

Note that this new behavior only applies to conferences being initiated from the telephone. The
original behavior of conferencing all calls still applies if the conference function is initiated from
elsewhere such as from an application like one-X Portal.
• Changing which call is currently highlighted
On phones with a set of cursor keys (four cursor keys around an OK key), the up and down
cursor key can be used to change the current highlighted call (or call appearance if idle).
This can be done even whilst there is a currently connected call. On touchscreen phones, the
cursor buttons on the right-hand edge of the screen can be used for the same purpose.
The method of highlighting is:
• - 1400/1600 Series Telephones - On these phones only details of a single call are shown
on the display at any time. The displayed call is the currently highlighted call.
- 9500/9600/J100 Series Telephones - On most phones in these series, the background
of the shading is changed for the currently selected call. The exceptions are 9611, 9621,
9641, J159 and J179 telephones where a yellow symbol is shown on the right of the
highlighted call.

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Conferencing

Related links
Conferencing on page 913

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Chapter 97: Ad-Hoc Conferencing

An ad-hoc conference is one created by the system on the fly. For example, when a user with two
calls in progress conferences those calls using their phone. For ad-hoc conferences, all internal
users are treated as moderators.
Related links
Dropping External Party Only Conferences on page 921
Adding Callers to an Ad-Hoc Conference on page 921

Dropping External Party Only Conferences

About this task
It may be desirable to prevent ad-hoc conferences from continuing if there are no internal users
involved. This can be enabled for the whole system.
Procedure
1. Select System Settings.
2. Click System.
3. Select Drop External Only Impromptu Conference.
• If enabled, when the last remaining internal user exits the conference, the conference is
ended regardless of whether it still includes other external parties
• If disabled, the conference is only ended when the last party exits the conference.
4. Click Update.
Related links
Ad-Hoc Conferencing on page 921

Adding Callers to an Ad-Hoc Conference

The method of starting an ad-hoc conference depends on the particular phone or softphone being
used. It will typically involve putting an existing call on hold, making an additional call and then

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Ad-Hoc Conferencing

selecting a conference option. The same method can usually be used for adding additional parties
to an existing conference.
If necessary, controls for starting and adding users to an ad-hoc conference can be created
using short codes and programmable buttons. Note that when used to add a party to an existing
conference, these controls also work with existing meet-me conferences.
Related links
Ad-Hoc Conferencing on page 921

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Chapter 98: Personal Meet-Me Conferences

Each user's own extension number is treated as their personal meet-me conference number. That
user is the conference’s only moderator. Other participants can join a personal meet-me conference
at any time, however the audio conference only starts when the owner also joins. If the user’s
optional conference PIN has been configured, the system prompts other callers for the PIN when
they try to access the personal meet-me conference.
• Participants who join a personal meet-me conference before the owner, are put on hold until
the owner joins. Whilst on-hold they hear repeated tones.
• If the user has a audio conference PIN set, callers joining the user’s personal meet-me
conference are prompted to enter that PIN.
• Personal and system meet-me features can create conferences that include only one or two
parties. These are still conferences that are using resources from the system's conference
capacity.
Related links
Setting a User’s Personal Conference PIN on page 923
Routing Internal Callers to a Meet-Me Conference on page 924
Routing External Callers to a Meet Me Conference on page 924
Personal Meet-Me Conference Callflow on page 925

Setting a User’s Personal Conference PIN

About this task
If the user has a audio conference PIN set, other callers attempting to joining their personal
meet-me conference are prompted to enter that PIN.
• Putting an L before the PIN, disables the user’s personal audio conference.
Procedure
1. From the menu bar, select Call Management and then Users.
2. Locate the user you want to edit and click the icon next to them.
3. On the User tab, select Audio Conference PIN.
4. Enter a numeric PIN code of up to 15 digits.
5. Click Update.

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Personal Meet-Me Conferences

Related links
Personal Meet-Me Conferences on page 923

Routing Internal Callers to a Meet-Me Conference

Internal users can join personal meet–me and system conferences using short codes or a
programmable button.
Using Short Codes
The Conference Meet Me short code feature can be used to create short codes that put the user
into a meet me conference.
The default short code for this is *66*N# where N is the conference ID of the conference required.
• Internal users can also use the short codes to transfer callers into a conference.
• The same short codes can also be used by external callers to join the conference by setting
the short code as the destination in features such as an auto-attendant transfer.
• For personal meet-me conferences, the short code can also specify a music source to
use rather than tones if the conference owner has not already joined. System meet-me
conferences use the conference’s own separate Hold Music setting.
Using a Programmable Button
The Conference Meet Me button feature can be used to create a programmable button to join a
personal meet–me or system conferences. The button can also be used to transfer other caller’s
into a conference.
• If the button is configured with a specific conference ID, the button also shows the status of
the conference.
• If the button is configured without a conference ID, when pressed the user is prompted to
enter the required conference ID.
This option is not supported on J139 and non-Avaya phones.
Related links
Personal Meet-Me Conferences on page 923

Routing External Callers to a Meet Me Conference

The same Conference Meet Me short codes used for internal callers (see Routing Internal Callers
to a Meet-Me Conference on page 924) can also be used for external callers.
Related links
Personal Meet-Me Conferences on page 923

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 Personal Meet-Me Conference Callflow

Personal Meet-Me Conference Callflow

The following flowcharts provide a simplified callflow for a personal meet-me conference.
Personal Meet-Me Conference

Get Owner’s Audio Conference PIN

If the conference owner has an Audio Conference PIN set, other participants are required to
enter that PIN in order to join the conference.

Join The Conference

Participants can join the conference ahead of the owner. However, if that is the case, they will hear
an announcement that the conference will not start until the moderator joins followed by tones or
music-on-hold. The conference starts once the owner also joins the conference.

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Personal Meet-Me Conferences

If the owner subsequently leaves the conference, the other participants hear ones or music-on-
hold again until the owner rejoins.
Related links
Personal Meet-Me Conferences on page 923

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Chapter 99: System Conferences

System meet-me conferences provide:

• Optional participant PIN.
• Multiple optional moderators based on listed internal users and/or caller’s who enter the
optional moderator PIN.
• Automatic conference prompts for access control in per-conference selectable language.
• Recording per system conference.

System Conference Examples

The system conference features allow the configuration of various different types of conference:
Conference Type Method
Simple Conference A conference with no PIN codes, no moderators.
Simple Conference with A simple conference with a PIN code required for entry but no moderators.
Access Control
Moderated Conference A conference which does not start until a listed moderator joins and ends when no
moderator remains.
Moderated Conference As above but with a PIN code required for entry.
with Access Control
External Moderated Using a moderator PIN to allow external callers to assume the moderator role.
Conference

Related links
Adding a System Conference on page 927
Editing a System Conference on page 928
Deleting a System Conference on page 928
System Conference Settings on page 929
Routing External Calls to a System Conference on page 931

Adding a System Conference

About this task
The number of system conferences that you can configure is limited as follows:

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System Conferences

Maximum Configured
IP500 V2 30
Other networks 120

This is in addition to the overall capacity limits for all conference types. See Conference
Capacities on page 915.
Procedure
1. Click Create a New Record
2. Configure the system conference settings. See System Conference Settings on
page 929.
3. Click Save.
Related links
System Conferences on page 927

Editing a System Conference

Procedure
1. Click on the settings field to edit the entry.
2. Configure the system conference settings. See System Conference Settings on
page 929.
3. Click Save.
Related links
System Conferences on page 927

Deleting a System Conference

About this task
• Before deleting an entry, check that it is not being used as the destination for any other
functions such as an auto-attendant action or incoming call route.
Procedure
1. Click on the Delete Current Record icon to delete the record.
2. Click Yes to confirm the deletion.
Related links
System Conferences on page 927

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 System Conference Settings

System Conference Settings

Conferences > Create a New Record
These settings are used to define the operation of a system meet-me conferences.
Field Description
Conference ID Range = Up to 15 digits.
This ID is shown in the destination list for auto-attendant actions and incoming call routes.
The ID can also be used with short code and programmable button features in order to
access the conference.
• Do not enter a number that matches a user’s extension number. Doing so will override
that user’s personal meet-me conference facility.
• It is advisable not to use conference ID’s that are near the range that may be in use for
ad-hoc conferences as above (100 plus). Once a conference ID is in use by an ad-hoc
conference, it is no longer possible to join the conference using the various conference
meet me features.
Name This is a short name to help indicate the system conferences intended use. For example,
“Sales Team”.
Moderator List Optional. Default = No moderators.
List the internal users who are moderators for this system conference, up to a maximum of
8 moderators. When set:
• The conference Hold Music is played to other participants when there is no moderator in
the conference.
• These user’s do not need to enter a PIN in order to access the conference.
• Listed users using the User Portal application can view the conference PIN details.
In addition:
• Other participants, including external participants, can become moderators by entering
the Moderator Pin when they join the conference.
• Conferences with no defined moderators (blank Moderator List and no Moderator Pin)
start immediately any caller joins and can have recording started/stopped by any internal
user.
Delegate Pin Optional. Range = Up to 30 digits.
If set, the system will prompt callers, other than those in the Moderator List list, to enter a
PIN before it allows them to join the conference.
The system allows 3 PIN entry attempts before disconnecting the caller.
Moderator Pin Optional. Range = Up to 30 digits.
If set, callers who enter this PIN rather than the Delegate Pin are added to the conference
as a moderator. This allows moderators who are not in the Moderator List including
external callers. Note however that external callers will not be able to access moderator
controls other than starting/stopping the conference by their presence.
Table continues…

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System Conferences

Field Description
Hold Music Default = Tone
If the conference has been configured with moderators, this music is played to other
participants who join the conference when no moderator is present. The music is also
played if any present moderators leave the conference.
• Tone – Play repeated system tones to participants whilst waiting for a conference
moderator.
• System – Use the system’s default music-on-hold. This option is only shown in a music-
on-hold file has been uploaded.
• If other music sources have been configured, they can also be selected from the drop-
down list.
Before the hold music is played, participants will hear a prompt informing them of the
reason for hearing the music.
Speech AI Default = Same as system
On subscription systems, this and other text-to-speech options are available if the System
| Voicemail setting for Google Speech AI is enabled.
• If set to Same as System, the settings of the System | Voicemail form are used for TTS
prompts.
• If set to Custom, the Language and Voice fields below can be used.
Language Default = Matches the system locale.
Set the language used for prompts provided by the system for the system conference.
Voice Sets the voice to be used with the speech language. The number of voices available varies
depending on the speech language selected.
Recording Type Default = Manual
Sets the method by which recording of the system conference is controlled:
• Manual – Recording can be started/stopped by moderators.
• Private – No recording allowed.
• Automatic – Automatically start recording the conference when started. The recording
can be stopped/restarted by moderators.
Recording Default = Conference Mailbox
Destination
Sets the destination for system conference recordings. Note that the selected option may
also affect the maximum recording length:
• Conference Mailbox - Place calls into a standard group mailbox, using the conference
ID as the mailbox number. Maximum recording length 60 minutes. Message waiting
indication and visual voice access can be configured by adding C<conference ID> to a
user’s source numbers.
• Conference VRL - Transfer the conference recordings to the systems VRL application
(on subscription systems, set by the System > System > Media Archival Solution
setting). Maximum recording length 5 hours.
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 Routing External Calls to a System Conference

Field Description
Meeting Arrival Default = Off
Announcement
If enabled, the system plays this prompt to callers before they join the conference. If
conference PIN codes have been defined, it is played before the request to the caller to
enter their PIN code.
• Audio Output – Use an uploaded audio file. See .The file must be a .wav file in Mono
PCM 16-bit format, either 8, 16 or 22KHz. Maximum length 10 minutes. To upload a file
click on Upload and select the required file. Alternatively, click and drag the file onto the
download box.

Note:
Use IP Office Web Manager to upload the .wav file.
• Text-to-Speech – Use a prompt generated using TTS. Up to 200 characters.

Related links
System Conferences on page 927

Routing External Calls to a System Conference

External callers can be routed to a conference using a number of methods:
• The conference ID appears as Conf:<id> in the Destination drop-down list for many
functions:
- From an Incoming Call Route, the Destinations drop-down includes system meet-me
conferences configured on the system.
- Through an auto-attendant, the configured conferences appear in the list of targets for
Unsupervised Transfer actions.
• Using the Dial By Conference action, callers routed to an auto-attendant can dial the
conference ID required.
• For other scenarios, the conference ID can be used as the number to which a call should be
routed using the format *<ID>#. For example in the telephone number field of a short code.
Related links
System Conferences on page 927

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Part 11: Short Codes

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Chapter 100: Short Code Overview

Whenever the system receives a set of digits to process, if those digits do not match a user or
group extension number, the system will look for a short code match. The matching short code then
defines what action (short code feature) should be applied to the call, where it should be routed and
which of the dialed digits, if any, should be used in the subsequent action.
This applies to digits dialed by a telephone user, sent by a user selecting a directory contact or
speed dial, and in some cases to digits received with an incoming call on a line.
This section provides an overview of short codes configuration and use.

Warning:
• The dialing of emergency numbers must not be blocked. Whenever short codes are
edited, you must ensure that the ability of users to dial emergency numbers is tested
and maintained. See Configuration for Emergency Calls on page 652.
Short Code Fields
Each short code has the following fields:
• Short Code: The digits which, if proved to be a best match to the dialed digits, trigger use of
the short code. In addition to the normal dialing digits (0 to 9 plus * and #), characters can also
be used as follows:
- Some characters have special meaning. For example, the wildcard X to match any single
digit or N to match any set of digits. See Short Code Characters on page 935
- Using characters also allows the creation of short codes which cannot be dialed from a
phone but can be dialed from some applications.
• Telephone Number: The number used by the short code feature if needed, for example the
outgoing number for a call to be passed to an external telephone line. Again special characters
can be used in this field, see Short Code Characters on page 935.
• Line Group ID: This field is used for short codes that result in a number to be dialed, that is
any short code set to one of the various Dial short code features. When that is the case, this
field specifies the outgoing line group or ARS form to be used for the call.
- For Dial Emergency short codes, this is overridden by the Emergency ARS setting of the
extension's Location if configured.
• Feature: This sets the action to performed by the short code. See Short Code Features on
page 953.
• Locale: Features that transfer the call to voicemail indicate the language required. If the
required set of language prompts is not available, the voicemail system will fallback to another
appropriate language if possible (refer to the appropriate voicemail installation manual for

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Short Code Overview

details). The locale sent to the voicemail server by the system is determined in the following
order of priority:
1. Short Code Locale: The short code locale, if set, is used if the call is routed to voicemail
using the short code.
2. Incoming Call Route Locale: The incoming call route locale, if set, is used if caller is
external.
3. User Locale: The user locale, if set, is used if the caller is internal.
4. System Locale: If no user or incoming call route locale is set, the system locale is used
unless overridden by a short code locale. Systems using Embedded Voicemail, if the
required set of upgraded language prompts to match the locale is not present on the
system SD card, Manager will display an error. The required prompt set can be uploaded
from Manager using the Add/Display VM Locales option.
• Force Account Code: When selected, if the short code results in the dialing of an external
number, the user is prompted to enter a valid account code before the call is allowed to
continue. See Account Code Configuration on page 722.
• Force Authorization Code: When selected, if the short results in the dialing of an external
number, the user is prompted to enter a valid authorization code before the call is allowed to
continue. See Configuring authorization codes on page 705.
Short Code Descriptions
The short method for describing short codes in this manual, for example 9N/Dial/./0, indicates the
settings of main short code fields, each separated by a / as follows:
• Code: In this case 9N.
• Feature: In this case Dial.
• Telephone Number: In this case the symbol . representing all dialed digits.
• Line Group ID: In this case the call is sent to outgoing line group 0.
Example Short Codes
• *17/VoicemailCollect/?U A user dialing *17 is connected to their own mailbox to collect
messages.
• *14*N#/FollowMeTo/N If a user dials *14*210# at their own extension, their calls are redirected
to extension 210.
Types of Short Code
In addition to different short code features, there are different types of short code:
• Dialing Short Codes: The following types of short code applied to on-switch dialing. The result
may be an action to be performed by the system, a change to the user's settings or a number
to be dialed. The order below is the order of priority in which they are used when applied to
user dialing.
- User Short Codes: These are usable by the specific user only. User short codes are applied
to numbers dialled by that user and to calls forwarded via the user.
- User Rights Short Codes: These are usable by any users associated with the user rights in
which they are set. User Rights short codes are only applied to numbers dialed by that user.
For example they are not applied to calls forwarded via the user.

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 Short Code Characters

- System Short Codes: These are available to all users on the system. They can be
overridden by user or user rights short codes.
• Post-Dialing Short Codes: When any the short code above result in a number to be dialed,
further short code can be applied to that number to be dialed. This is done using the following
types of short codes.
- ARS (Alternate Route Selection) Short Codes: The short code that matches dialing can
specify that the resulting number should be passed to an ARS form. The ARS form can
specify which routes should be used for the call by using further short code matches and
also provide option to use other ARS forms based on other factors such as time and
availability of routes.
- Transit Network Selection (TNS) Short Codes: Used on T1 ISDN trunks set to use AT&T
as the Provider. Applied to the digits presented following any other short code processing.
• Incoming Number Short Codes: On certain types of trunks short codes can be applied to the
incoming digits received with calls.
- Line Short Codes: These short codes are used to translate incoming digits received with
calls. The stage at which they are applied varies between different line types and may be
overridden by an extension number match.
Related links
Short Code Characters on page 935
User Dialing on page 940
Application Dialing on page 942
Secondary Dial Tone on page 942
? Short Codes on page 944
Short Code Matching Examples on page 944
Default System Short Code List on page 947

Short Code Characters

The short code fields Short Code and Telephone Number can contain the normal dialable digits
*, # and 0 to 9. In addition they can also use a range of special characters as listed below.
Short Code Field Characters
Description
? Default Match
This character can be used on its own to create a short code match in the absence of any other
short code match.
?D Default Number Dialing
This character combination makes a call to the defined phone number as soon as the user goes
off-hook.
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Short Code Overview

Description
?D(t) Default Number Dialing Timeout
The character x represents time in seconds. If a phone is off-hook or speaker is enabled and no
number is dialed for t seconds, the phone makes a call to the defined phone number. A maximum
of 30 seconds is used for t though system accepts values more than 30 on the interface.
F Failed Authentication
Match incoming SIP calls which failed authentication. See SIP Calling Number Verification (STIR/
SHAKEN) on page 870.
N Match Any Digits
Matches any dialed digits (including none). The Dial Delay Time or a following matching character
is used to resolve when dialing is complete.
P Authenticated
Match incoming SIP calls which were authenticated. The character can be followed by the
required attestation level or levels in " " quote marks. See SIP Calling Number Verification (STIR/
SHAKEN) on page 870.
Q Unauthenticated
Match incoming SIP calls which were not authenticated. See SIP Calling Number Verification
(STIR/SHAKEN) on page 870.
X Match a Digit
Matches a single digit. When a group of X's is used, the short code matches against the total
number of X's.
[] Secondary Dial Tone Trigger
For pre-4.0 IP Office systems used to trigger secondary dial tone. Not used for Release 4.0+. See
Secondary Dial Tone on page 942.
; Receive Sending Complete
When used this must be the last character in the short code string.
• If the Dial Delay Count is 0, a ; instructs the system to wait for the number to be fully dialed,
using the Dial Delay Time or the user dialing #, to indicate completion and then acting on the
short code.
• If the Dial Delay Count is not zero, the dialing is only evaluated when # is pressed. The
majority of North-American telephony services use en-bloc dialing. Therefore the use of a ; is
recommended at the end of all dialing short codes that use an N before routing those calls to a
trunk or ARS. This is also recommended for all dialing where secondary dial tone short codes are
being used.

Telephone Number Field Characters

Description
A Allow Outgoing CLI
Allow the calling party number sent with the call to be used. This character may be required by
service providers in some locales.
Table continues…

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 Short Code Characters

Description
C Use Called Number Field Place any following digits in the outgoing call's Called number field
rather than the Keypad field.
D Wait for Connect
Wait for a connect message before sending any following digits as DTMF.
E Extension Number
Replace with the extension number of the dialing user. Note that if a call is forwarded this will be
replaced with the extension number of the forwarding user.
h Hold Music Source
When used as part of the short code telephone number field, this character allows the source
for music on hold to be selected. Enter h(X)where X indicates the required hold music source if
available. This overrides any previous hold music selection that may have been applied to the call.
• For IP500 V2 systems, the value of X can be 1 to 4.
• For systems on a Linux based server, the value of X can be 1 to 32.
• When used with Park Call short codes, the h(X) should be entered before the park slot number
part of the telephone number.
I Use Information Packet
Send data in an Information Packet rather than Set-up Packet.
K Use Keypad Field
Place any following digits in the outgoing call's Keypad field rather than the Called Number field.
Only supported on ISDN and QSIG.
l Last Number Dialed (lower case L)
Use the last number dialed.
L Last Number Received
Use the last number received.
N Dialed Digit Wildcard Match
Substitute with the digits used for the N or X character match in the Short Code number field.
Table continues…

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Short Code Overview

Description
p Priority
The priority of a call is normally assigned by the Incoming Call Route or else is 1-Low for all other
calls. Dial Extn short codes can use p(x) as a suffix to the Telephone Number to change the
priority of a call. Allowable values for x are 1, 2 or 3 for low, medium or high priority respectively.
In situations where calls are queued, high priority calls are placed before calls of a lower priority.
This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where Voicemail Pro is being
used to provided queue ETA and queue position messages to callers since those values will no
longer be accurate when a higher priority call is placed into the queue. Note also that Voicemail
Pro will not allow a value already announced to an existing caller to increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt group's Queue
Length Limit, the limit is temporarily raised by 1. This means that calls already queued are not
rerouted by the addition of a higher priority call into the queue.
r Ring Tone Plan
When used as part of the short code telephone number field, this character can specify a Ring
Tone Plan number. Enter r(X)where X is 1 to 8 indicating the Ring Tone Plan number to use.
S Calling Number
Place any following digits into the outgoing call's calling number field. Using S does not alter any
allow or withhold CLI setting associated with the call, the short code characters A or W should be
used respectively.
• On mobile twinned calls, if the original party information is used or a specific calling party
information CLI is set, that number overrides setting the outgoing CLI using short codes.
• Note that for SIP trunks, the SIP URI configuration options override this setting.

Warning:
• Changing the outgoing CLI for calls requires the line provider to support that function. You
must consult with your line provider before attempting to change the outgoing CLI, failure
to do so may result in loss of service. If changing the outgoing CLI is allowed, most line
providers require that the outgoing CLI used matches a number valid for return calls on the
same trunks. Use of any other number may cause calls to be dropped or the outgoing CLI
to be replaced with a valid number. On mobile twinned calls, if the original party information
is used or a specific calling party information CLI is set, that number overrides setting the
outgoing CLI using short codes.
SS Pass Through Calling Number
Pass through the Calling Party Number. For example, to provide the incoming ICLID at the far end
of a VoIP connection, a short code ? with telephone number .SS should be added to the IP line.
i National
Both the S and SS characters can be followed by an i, that is Si and SSi. Doing this sets the calling
party number plan to ISDN and number type to National. This may be required for some network
providers.
Table continues…

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 Short Code Characters

Description
t Allowed Call Duration
Set the maximum duration in minutes for a call plus or minus a minute. Follow the character with
the number of minutes in brackets, for example t(5).
U User Name
Replace with the User Name of the dialing user. Used with voicemail.
W Withhold Outgoing CLI
Withhold the sending of calling ID number. Operation is service provider dependent.
Y Wait for Call Progress Message
Wait for a Call Progress or Call Proceeding message before sending any following digits as DTMF.
For example, the Y character would be necessary at a site where they have signed up with their
telephone service provider to withhold international dialing until a DTMF pin/account number is
entered that initiates the call progress/proceeding message.
Z Calling Party Name
This option can be used with trunks that support the sending of name information. The Z character
should be followed by the name enclosed in " " quotation marks. Note that their may be name
length restrictions that vary between line providers. The changing of name information on calls
being forwarded or twinned may also not be supported by the line provider.
@ Use Sub Address Field
Enter any following digits into the sub-address field.
. Dialed Digits
Replace with the full set of dialed digits that triggered the short code match.
, One Second Pause
Add a one second pause in DTMF dialing.
; Receive Sending Complete
When used this must be the last character in the short code string. If the Dial Delay Count is 0, a ;
instructs the system to wait for the number to be fully dialed, using the Dial Delay Time or the user
dialing #, to indicate completion and then acting on the short code. If the Dial Delay Count is not
zero, the dialing is only evaluated when # is pressed.
"" Non-Short Code Characters
Use to enclose any characters that should not be interpreted by the IP Office as possible
short code special characters. For example, characters being passed to the voicemail server for
interpretation there.
• Ensure you use straight quotation marks like "…" when entering short codes into the IP Office
configuration. Various editing, publishing and copying tools often replace those with angled or
smart-quotes such as “…‟.

Related links
Short Code Overview on page 933

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Short Code Overview

User Dialing
The following rules are used when short code matching is performed for user dialing:
• A short code is used immediately an exact match is found unless followed by a ; semi-colon.
- If a ; semi-colon is present, dialing complete can be indicated by the user pressing # or the
Dial Delay Time (see below) expiring.
• If no match is found but partial matches exist, the user can continue dialing.
• If no match or partial matches are found, incompatible is returned.
• The following precedence is used to determine which short codes are used:
- Extension number matches override all short codes.
- User short codes override user rights and system short codes.
- User Rights short code matches override system short codes.
• When multiple exact matches exist:
- The match with the most specified digits rather than wildcards is used.
- If there are still more than one match, the match with the most exact length is used. This
means X single-digit wildcards will override N multiple0digit wildcards when both match.
• The rules above are applied even if the number is dialed by selection from a directory or
using any other method of stored number dialing.
User Digit Dialing Settings
The following system settings influence user dialing.
• Dial Delay Count: Default = 0 (US/Japan), 4 (ROW).
This value sets the number of digits dialed before the system starts looking for short code
matches.
• Dial Delay Time: Default = 4 seconds (US/Japan), 1 second (ROW).
This value sets the maximum allowed interval between the dialing of each digit. If exceeded,
the system treats dialing as completed and looks for a short code match even if the Dial
Delay Count has not been reached.
• Off-Hook Timer:
When a user goes off-hook, the system starts a 30 second off-hook timer (10 seconds
in Italy). If the off-hook timer expires before a short code match occurs, the user is
disconnected.

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 User Dialing

User Dialing Flowchart

Related links
Short Code Overview on page 933

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Short Code Overview

Application Dialing
Numbers speed dialed by system applications such as SoftConsole are treated differently. Since
the digits are received en bloc as a single group, they can override some short code matches.
The same applies to short codes used within system configuration settings such as Incoming Call
Route destinations.
Example:
• Telephone Number: 12345678
• Short Code 1: 1234XX/Dial/Extn/207
• Short Code 2: 12345678/Dial Extn/210
If dialed manually by the user, as soon as they have dialed 123456 a match to short code 1
occurs. They can never dial short code 2.
If dialed using an application, 12345678 is sent as a string and a match to short code 2 occurs.
Partial Dialing
If the application dialing does not trigger an exact match, the user can dial additional digits through
their extension. The processes for normal user dialing are applied.
Non-Digit Short Codes
Short codes can be created that use alphabetic characters instead of numbers. While these short
codes cannot be dialed from a phone, they can be dialed using application speed dials and
settings. However characters that are interpreted as special short code characters will still be
interpreted as such.
Related links
Short Code Overview on page 933

Secondary Dial Tone

Some locales prefer to provide users with secondary dial tone once they have started dialing
external calls. This dial tone is heard by the user until they have completed dialing and a trunk
is seized at which point call progress tones are provided by the trunk, or camp on/busy tone is
provided by the system if the required trunk cannot be seized.
Release 4.0 and Higher
The use of secondary dial tone is provided through the Secondary Dial Tone check box option
on the ARS form to which the call is routed. When on, this setting instructs the system to play
secondary dial tone to the user.
The tone used is set as either System Tone (normal dial tone) or Network Tone (secondary dial
tone). Both tone types are generated by the system in accordance with the system specific locale
setting. Note that in some locales normal dial tone and secondary dial tone are the same.

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 Secondary Dial Tone

When Secondary Dial Tone is selected, the ARS form will return tone until it receives digits with
which it can begin short code matching. Those digits can be the result of user dialing or digits
passed by the short code which invoked the ARS form. For example with the following system
short codes:
In this example, the 9 is stripped from the dialed number and is not part of the telephone number
passed to the ARS form. So in this case secondary dial tone is given until the user dials another
digit or dialing times out.
• Code: 9N
• Telephone Number: N
• Line Group ID: 50 Main
In this example, the dialed 9 is included in the telephone number passed to the ARS form. This will
inhibit the use of secondary dial tone even if secondary dial tone is selected on the ARS form.
• Code: 9N
• Telephone Number: 9N
• Line Group ID: 50 Main
Pre-4.0 IP Office Secondary Dial Tone
Pre-4.0 systems provided dial tone through the use of the short code feature Secondary Dial Tone
and the [ ] special characters. For example, on a system where 9 is used as a prefix for external
dialing, the system short code 9/./Secondary Dial Tone/0 will trigger secondary dial tone when
users dial a number prefixed with 9. This method is not supported by Release 4.0 which provides
ARS forms for the control of outgoing calls.
In order to allow further digit matching, the digits dialed are put back through short code matching
against any short codes that start with [n] where n is the digit used to trigger the system secondary
dial tone short code.
On all systems where secondary dial tone is used, a ; should also be used in dialing short codes
that contain N.
For example:
System Short Codes
• 9/SecondaryDialTone/.
• [9]0N;/Dial/0
User Short Code
[9]0N;/Busy/0
The user dials 90114445551234. The 9 is matches the system secondary dial tone short code
and unlike other short codes this is applied immediately. The user's dialing is put through short
code matching again using the normal order of precedence but matched to possible short codes
beginning [9]. In this case the user's [9]0N; short code would take precedence over the system
[9]0N; short code.

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Short Code Overview

Related links
Short Code Overview on page 933

? Short Codes
The ? character can be used in short codes in the following ways:
Default Short Code Matching:
? short codes are used in short code matching in the following way. If no user or system short
code match is found, the system will then look for a ? short code match. It will look first for a user ?
short code and then, if not found, a system ? short code.
Example: On systems outside North America, the system short code ?/Dial/./0 is added as a
default short code. This short code provides a match for any dialing to which there is no other
match. Therefore, on systems with this short code, the default is that any unrecognized number
will be dialed to Outgoing Line Group 0.
Hot-Line Dialing:
A user short code ?D can be used to perform a short code action immediately the user extension
goes off-hook. This is supported with Dial type short code features. Typically it is used with door,
lift and lobby phones to immediately connect the phone to a number such as the operator or
reception.
Voicemail Collect Short Codes:
The ? character can appear in the Telephone Number field of a short code. This is done with
short codes using the VoicemailCollect feature. In this instance the ? character is not interpreted
by the system, it is used by the voicemail server.
Related links
Short Code Overview on page 933

Short Code Matching Examples

The following examples are not meant as practical examples. However they are simple to
implement and test on real system without conflicting with its normal operation. They illustrate
the interaction between different short codes in resolving which short code is an exact match.
They assume that extension numbers are in the 200 to 299 range.
• The term 'dials' means dialing the indicated digit or digits without the inter-digit Dial Delay
Time expiring.
• The term 'pause' means a wait that exceeds the inter-digit Dial Delay Time.

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 Short Code Matching Examples

Scenario 1
• Short Code 1 = 60/Dial Extn/203
• Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 No possible match, incompatible returned immediately
2 6 No exact match but there is a potential match, so the system waits. When the Dial
Delay Time expires, no exact match is found so incompatible is returned.
3 60 Exact match to Short Code 1. Extension 203 called immediately.
4 61 No possible match, the system returns incompatible.

Scenario 2
• Short Code 1 = 60/Dial Extn/203
• Short Code 2 = 601/Dial Extn/210
• Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 No possible match, incompatible returned immediately
2 60 Exact match to Short Code 1. Extension 203 called immediately.
3 601 Exact match to Short Code 1 as soon as the 0 is dialed. The user cannot manually
dial 601.

Scenario 3
Short Code 1 = 60/Dial Extn/203
Short Code 2 = 601/Dial Extn/210
Dial Delay Count = 3. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 Insufficient digits to trigger matching. The system waits for additional digits or for
Dial Delay Time to expire. When Dial Delay Time expires, no possible match is
found so incompatible is returned.
2 60 Insufficient digits to trigger matching. The system waits for additional digits or for
Dial Delay Time to expire. When Dial Delay Time expires, matching started and
exact match to Short Code 1 occurs. .
3 601 Third digit triggers matching. Exact match to Short Code 2. Extension 210 dialed
immediately.
4 60# # is treated as a digit and as the third digit triggers matching. No exact match
found. The system returns incompatible.

Scenario 4
• Short Code 1 = 60;/Dial Extn/203
• Short Code 2 = 601/Dial Extn/210

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Short Code Overview

• Dial Delay Count = 3. Dial Delay Time = 4 seconds.

Test Dialing Effect
1 8 Insufficient digits to trigger matching. The system waits for additional digits or for
Dial Delay Time to expire. When Dial Delay Time expires, no possible match is
found so incompatible is returned.
2 6 Insufficient digits to trigger matching. The system waits for additional digits or for
the interdigit Dial Delay Time to expire. If the Dial Delay Time expires, a potential
match exists to a short code that uses ; so the system waits for an additional digit
until the off-hook timer expires.
3 60 As above but an additional digit now may create a match.
If 1 is dialed, it creates an exact match to Short Code 2 and is used immediately.
If 0, * or 2 to 9 is dialed, no possible match exists. The system returns
incompatible.
If the next digit is a #, it is treated as signaling dialing complete rather than being a
digit. Short code 1 becomes an exact match and is used immediately.
4 601 Third digit triggers matching. Exact match to Short Code 2. Extension 210 dialed
immediately.

Scenario 5
• Short Code 1 = 601/Dial Extn/203
• Short Code 2 = 60N/Dial Extn/210
• Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 6 No exact match but there is a potential match, so the system waits for additional
dialing. If the Dial Delay Time expires, no exact match is found so incompatible is
returned.
2 60 Potential match to both short codes. The system waits for additional dialing. If the
Dial Delay Time expires, Short Code 2 becomes an exact match with N blank.
3 601 Exact match to Short Code 1. Used immediately
4 602 Exact match to Short Code 2. Used immediately.

Scenario 6
• Short Code 1 = 601/Dial Extn/203
• Short Code 2 = 60N/Dial Extn/210
• Short Code 3 = 60X/Dial Extn/207
• Dial Delay Count = 0. Dial Delay Time = 4 seconds.

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 Default System Short Code List

Test Dialing Effect

1 6 No exact match but there are potential matches so the system waits for
additional dialing. If the Dial Delay Time expires, no exact match has occurred
so incompatible is returned.
2 60 Potential match to all short codes. System waits for additional dialing. If the Dial
Delay Time expires, Short Code 2 becomes an exact match with N blank. If a digit
is dialed, Short Code 3 becomes a more exact match and is used.
3 601 Exact match all short code, however Short Code 1 is treated as being more exact
(more matching digits) and is used immediately
4 602 Exact match to short codes 2 and 3, however the Short Code 3 is treated as being
more exact (length match) and is used immediately.

Scenario 7
• Short Code 1 = 601/Dial Extn/203
• Short Code 2 = 60N/Dial Extn/210
• Short Code 3 = 6XX/Dial Extn/207
• Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 6 No exact match but there are potential matches so the system waits for
additional dialing. If the Dial Delay Time expires, no exact match has occurred
so incompatible is returned.
2 60 Potential match to all short codes. System waits for additional dialing. If the Dial
Delay Time expires, Short Code 2 becomes an exact match with N blank. If a digit
is dialed, Short Code 3 becomes an more exact match and is used.
3 601 Exact match all short code, however Short Code 1 is treated as being more exact
(more matching digits) and is used immediately
4 602 Exact match to short codes 2 and 3, however the Short Code 2 is treated as being
more exact (more matching digits) and is used immediately.
5 612 Exact match to Short Code 3.

Related links
Short Code Overview on page 933

Default System Short Code List

Most control units are available in A-Law and U-Law models. Typically U-Law models are supplied
to North American locales, A-Law models are supplied to the rest of the world. In addition to the
using different default companding for digital lines and phone, A-Law and U-Law models support
different default short codes. The following table lists the default system short codes present in a
system's configuration.

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Short Code Overview

Standard Mode
Short Code Telephone Number Feature A-Law U-Law
*00 Blank Cancel All Forwarding
*01 Blank Forward Unconditional On
*02 Blank Forward Unconditional Off
*03 Blank Forward On Busy On
*04 Blank Forward On Busy Off
*05 Blank Forward On No Answer On
*06 Blank Forward On No Answer Off
*07*N# N Forward Number
*08 Blank Do Not Disturb On
*09 Blank Do Not Disturb Off
*10*N# N Do Not Disturb Exception Add
*11*N# N Do Not Disturb Exception Del
*12*N# N Follow Me Here
*13*N# N Follow Me Here Cancel
*14*N# N Follow Me To
*15 Blank Call Waiting On
*16 Blank Call Waiting Off
*17 ?U Voicemail Collect
*18 Blank Voicemail On
*19 Blank Voicemail Off
*20*N# N Set Hunt Group Night Service
*21*N# N Clear Hunt Group Night Service
*22*N# N Suspend Call
*23*N# N Resume Call
*24*N# N Hold Call
*25*N# N Retrieve Call
*26 Clear CW
*27*N# N Hold CW
*28*N# N Suspend CW
*29 Blank Toggle Calls
*30 Blank Call Pickup Any
*31 Blank Call Pickup Group
*32*N# N Call Pickup Extn
*33*N# N Call Queue
Table continues…

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 Default System Short Code List

Short Code Telephone Number Feature A-Law U-Law

*34N; N Hold Music
*35*N# N Extn Login
*36 Blank Extn Logout
*37*N# N Call Park
*38*N# N Unpark Call
*39 1 Relay On
*40 1 Relay Off
*41 1 Relay Pulse
*42 2 Relay On
*43 2 Relay Off
*44 2 Relay Pulse
*45*N# N Acquire Call
*46 Blank Acquire Call
*47 Blank Conference Add
*48 Blank Voicemail Ringback On
*49 Blank Voicemail Ringback Off
*50 Blank Forward Huntgroup On
*51 Blank Forward Huntgroup Off
*52 Blank Cancel or Deny
*53*N# N Call Pickup Members
*55 Blank Stamp Log
*57*N# N Forward On Busy Number
*70 Blank Call Waiting Suspend
*70*N# N Dial Physical Extn by Number
*71*N# N Dial Physical Extn by Id
*9000* "MAINTENANCE" Relay On
*91N; N".1" Record Message
*92N; N".2" Record Message
*99; "edit_messages" Voicemail Collect
9N N Dial
? . Dial

Server Edition
Short Code Telephone Number Feature A-Law U-Law
*00 Blank Cancel All Forwarding
Table continues…

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Short Code Overview

Short Code Telephone Number Feature A-Law U-Law

*01 Blank Forward Unconditional On
*02 Blank Forward Unconditional Off
*03 Blank Forward On Busy On
*04 Blank Forward On Busy Off
*05 Blank Forward On No Answer On
*06 Blank Forward On No Answer Off
*07*N# N Forward Number
*08 Blank Do Not Disturb On
*09 Blank Do Not Disturb Off
*10*N# N Do Not Disturb Exception Add
*11*N# N Do Not Disturb Exception Del
*12*N# N Follow Me Here
*13*N# N Follow Me Here Cancel
*14*N# N Follow Me To
*17 ?U Voicemail Collect
*18 Blank Voicemail On
*19 Blank Voicemail Off
*20*N# N Set Hunt Group Night Service
*21*N# N Clear Hunt Group Night Service
*29 Blank Toggle Calls
*30 Blank Call Pickup Any
*31 Blank Call Pickup Group
*32*N# N Call Pickup Extn
*33*N# N Call Queue
*34N; N Hold Music
*35*N# N Extn Login
*36 Blank Extn Logout
*37*N# N Call Park
*38*N# N Unpark Call
*44 2 Relay Pulse
*45*N# N Acquire Call
*46 Blank Acquire Call
*47 Blank Conference Add
*48 Blank Voicemail Ringback On
*49 Blank Voicemail Ringback Off
Table continues…

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 Default System Short Code List

Short Code Telephone Number Feature A-Law U-Law

*50 Blank Forward Huntgroup On
*51 Blank Forward Huntgroup Off
*52 Blank Cancel or Deny
*53*N# N Call Pickup Members
*55 Blank Stamp Log
*57*N# N Forward On Busy Number
*66*N# N Conference Meet Me
*70 Blank Call Waiting Suspend
*70*N# N Dial Physical Extn by Number
*71*N# N Dial Physical Extn by Id
*99; "edit_messages" Voicemail Collect
9N N Dial [1]
? . Dial [1]

Embedded Voicemail
The following additional short codes are automatically added when an auto-attendant is cadded to
the configuration.
Short Code Telephone Number Feature: Auto Attendant
*81XX "AA:"N".1" These short codes correspond to the morning, afternoon, evening
*82XX "AA:"N".2" and menu actions prompts respectively.

*83XX "AA:"N".3" When dialed, the value XX is replaced with the auto-attendant
number.
*84XX "AA:"N".4"
*87XX "AA:"N".7" This short code is used on systems using a Voicemail Pro auto-
attendant to record the no match prompt.
*800XX "AA:"N".00 These short codes are used to record prompts for Park and Page
*801XX "AA:"N".01 actions. Each short code corresponds to the different key to which
the action might be assigned, from 0 to 9, * and # respectively.
*802XX "AA:"N".02
When dialed, the value XX is replaced with the auto-attendant
*803XX "AA:"N".03 number.
*804XX "AA:"N".04
*805XX "AA:"N".05
*806XX "AA:"N".06
*807XX "AA:"N".07
*808XX "AA:"N".08
*809XX "AA:"N".09
*850XX "AA:"N".10
*851XX "AA:"N".11

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Short Code Overview

General
For U-Law systems, a 9N is a default short code on the Primary Server while a ? short code is a
default on all other servers.
Additional short codes of the form *DSSN, *SDN, *SKN, these are used by the system for internal
functions and should not be removed or altered. Short codes *#N and **N may also visible, these
are used for ISDN functions in Scandinavian locales.
The default *34 short code for music on hold has changed to *34N;.
Related links
Short Code Overview on page 933

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Chapter 101: Short Code Features

The following descriptions cover all short code features. However, the short codes available on a
system depend on the system type and software release of that system.
Related links
Auto Attendant on page 956
Auto Intercom Deny Off on page 957
Auto Intercom Deny On on page 957
Break Out on page 958
Barred on page 958
Busy On Held on page 959
Call Intrude on page 960
Call Listen on page 960
Call Park on page 962
Call Park and Page on page 962
Call Pickup Any on page 963
Call Pickup Extn on page 963
Call Pickup Group on page 964
Call Pickup Line on page 965
Call Pickup Members on page 965
Call Pickup User on page 966
Call Queue on page 966
Call Record on page 967
Call Steal on page 968
Call Waiting On on page 969
Call Waiting Off on page 969
Call Waiting Suspend on page 970
Cancel All Forwarding on page 970
Cancel Ring Back When Free on page 971
Change Login Code on page 972
Clear After Call Work on page 972
Clear Call on page 973
Clear CW on page 973
Clear Hunt Group Night Service on page 974
Clear Hunt Group Out Of Service on page 975

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Short Code Features

Clear Quota on page 975

Coaching Intrusion on page 976
Conference Add on page 976
Conference Meet Me on page 977
CW on page 978
Dial on page 979
Dial 3K1 on page 980
Dial 56K on page 980
Dial 64K on page 981
Dial CW on page 981
Dial Direct on page 982
Dial Direct Hot Line on page 982
Dial Emergency on page 983
Dial Extn on page 983
Dial Fax on page 984
Dial Inclusion on page 984
Dial Paging on page 985
Dial Physical Extension by Number on page 986
Dial Physical Extension By ID on page 986
Dial Speech on page 987
Dial V110 on page 987
Dial V120 on page 988
Dial Video on page 988
Disable ARS Form on page 988
Disable Internal Forwards on page 989
Disable Internal Forward Unconditional on page 989
Disable Internal Forward Busy or No Answer on page 990
Display Msg on page 990
Do Not Disturb Exception Add on page 991
Do Not Disturb Exception Delete on page 992
Do Not Disturb On on page 993
Do Not Disturb Off on page 993
Enable ARS Form on page 994
Enable Internal Forwards on page 994
Enable Internal Forward Unconditional on page 995
Enable Internal Forward Busy or No Answer on page 995
Extn Login on page 995
Extn Logout on page 997
Flash Hook on page 997
FNE Service on page 998
Follow Me Here on page 998

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 Follow Me Here Cancel on page 999
Follow Me To on page 999
Forward Hunt Group Calls On on page 1000
Forward Hunt Group Calls Off on page 1001
Forward Number on page 1001
Forward On Busy Number on page 1002
Forward On Busy On on page 1003
Forward On Busy Off on page 1003
Forward On No Answer On on page 1004
Forward On No Answer Off on page 1004
Forward Unconditional On on page 1005
Forward Unconditional Off on page 1005
Group Listen Off on page 1006
Group Listen On on page 1006
Headset Toggle on page 1007
Hold Call on page 1007
Hold CW on page 1008
Hold Music on page 1009
Hunt Group Disable on page 1009
Hunt Group Enable on page 1010
Last Number Redial on page 1010
MCID Activate on page 1011
Mobile Twinned Call Pickup on page 1011
Off Hook Station on page 1012
Outgoing Call Bar Off on page 1012
Outgoing Call Bar On on page 1013
Private Call Off on page 1014
Private Call On on page 1014
Priority Call on page 1015
Record Message on page 1015
Relay On on page 1016
Relay Off on page 1017
Relay Pulse on page 1017
Resume Call on page 1018
Retrieve Call on page 1019
Ring Back When Free on page 1019
Secondary Dial Tone on page 1020
Set Absent Text on page 1020
Set Account Code on page 1022
Set Authorization Code on page 1022
Set Fallback Twinning Off on page 1023

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Short Code Features

Set Fallback Twinning On on page 1023

Set Hunt Group Night Service on page 1023
Set Hunt Group Out Of Service on page 1024
Set Inside Call Seq on page 1025
Set Mobile Twinning Number on page 1026
Set Mobile Twinning On on page 1026
Set Mobile Twinning Off on page 1026
Set No Answer Time on page 1027
Set Outside Call Seq on page 1027
Set Ringback Seq on page 1028
Set Time Profile on page 1029
Set Wrap Up Time on page 1030
Speed Dial on page 1031
Shutdown Embedded Voicemail on page 1031
Stamp Log on page 1032
Startup Embedded Voicemail on page 1032
Suspend Call on page 1033
Suspend CW on page 1033
Start After Call Work on page 1034
Toggle Calls on page 1034
Unpark Call on page 1035
Voicemail Collect on page 1035
Voicemail Node on page 1037
Voicemail On on page 1037
Voicemail Off on page 1038
Voicemail Ringback On on page 1038
Voicemail Ringback Off on page 1039
Whisper Page on page 1040

Auto Attendant
This feature is used with auto-attendants for recording greetings and to transfer calls to an
auto-attendant.
Details
• Telephone Number:
- System short codes (*81XX, *82XX, *83XX and *84XX) are automatically added for
use with all auto-attendants. These are used for morning, afternoon, evening and menu
options greetings respectively. These short codes use a Telephone Number of the form

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 Auto Intercom Deny Off

"AA:"N".Y" where N is the replaced with the auto attendant number dialed and Y is 1, 2,
3 or 4 for the morning, afternoon, evening or menu option greeting.
- To add a short code to call an auto-attendant, omit the XX part. For example, add the short
code *80XX/Auto Attendant/"AA:"N if internal dialed access to auto-attendants is
required.
- System short codes *800XX, *801XX, …, *809XX, *850XX, and *851XX are also
automatically added for recording prompts for any Page and Page actions. The codes
correspond to the key to which the action has been assigned; 0 to 9, * and #
respectively. These short codes use a Telephone Number of the form "AA:"N".00",
…, "AA:"N".01", "AA:"N".10" and "AA:"N".11" respectively.
• Release: 2.0+.
• Programmable Button Control:
• Default Short Code: See Configuration Settings | Auto Attendant.
Related links
Short Code Features on page 953

Auto Intercom Deny Off

Details
• Telephone number:
• Default short code:
• Programmable Button Control: Auto Intercom Deny Off
Related links
Short Code Features on page 953

Auto Intercom Deny On

Details
• Telephone number:
• Default short code:
• Programmable Button Control: Auto Intercom Deny On
Related links
Short Code Features on page 953

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Short Code Features

Break Out
This feature is usable within a system multi-site network. It allows a user on one system in the
network to specify that the following dialing be processed by another system on the network as if
the user dialed it locally on that other system.
Details
• Telephone Number: The IP Address or Name of the system, using * characters in place of .
characters.
• Default Short Code:
• Programmable Button Control: BkOut
• Release: 4.0+.
Examples
On a system, to break out via a system called RemoteSwitch with the IP Address 192.168.42.3,
either of the following short codes could be used.
Example 1 allows break out using any remote switch by dialing its IP address, for example
*80*192*168*42*3#. Example 2 does this for a specific remote system by dialing just *81.
• Example 1
- Feature: Break Out
- Telephone Number: N
- Code: *80*N#
• Example 2
- Code: *81
- Telephone Number: RemoteSwitch
- Feature: Break Out
Related links
Short Code Features on page 953

Barred
This short code feature can be used for call barring by using the short code as the call destination.
This short code feature was previously called Busy. It has been renamed but its function has not
changed.
When used in an ARS form that has been configured with an Alternate Route, for callers whose
dialing has matched the short code no further routing is applied.
Details
• Telephone Number:

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 Busy On Held

• Default Short Code:

• Programmable Button Control:
• Release: 1.0+.
Related links
Short Code Features on page 953

Busy On Held
When on, busy on held returns busy to new calls when the user has an existing call on hold. This
short code feature is useful when a user does not want to be distracted by an additional incoming
call when they have a call on hold.
Details
• Telephone Number: Y or 1 for on, N or 0 for off.
• Default Short Code:
• Programmable Button Control: BusyH
• Release: 1.0+.
Example: Turning Busy on Held on
If on, when the user has a call on hold, new calls receive busy tone (ringing if analog) or are
diverted to Voicemail if enabled, rather than ringing the user.
This overrides call waiting when the user has a call on hold.
• Short Code: *12
• Telephone Number: Y
• Feature: BusyOnHeld
Example: Turning Busy on Held off
Another short code must be created to turn the Busy on Held feature off. If off, when the uses has
a call on hold, new calls will still get directed to the user.
• Short Code: *13
• Telephone Number: N
• Feature: BusyOnHeld
Related links
Short Code Features on page 953

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Short Code Features

Call Intrude
This feature allows you to intrude on the existing connected call of the specified target user. All call
parties are put into a conference and can talk to and hear each other. A Call Intrude attempt to a
user who is idle becomes a Priority Call.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
• Users can use privacy features to set a call cannot be intruded on and recorded.
• Intruding onto a user doing silent monitoring (see Call Listen on page 960) is turned into a
silent monitoring call.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Telephone Number: Target extension number.
• Default Short Code:
• Programmable Button Control: Intru
• See also: Call Listen on page 960, Coaching Intrusion on page 976, Dial Inclusion on
page 984, Whisper Page on page 1040.
• Release: 1.0+.
Related links
Short Code Features on page 953

Call Listen
This feature allows you to monitor another user's call without being heard.
• By default, monitoring is accompanied by a tone heard by all parties. Use of the tone is
controlled by the Beep on Listen setting on the System > Telephony > Tones and Music
tab.
Warning:
• Listening to a call without the other parties being aware is subject to local regulations.
You must ensure that you have complied with the local regulations. Failure to do so can
result in penalties.
The use of call listen is dependent on:
• The target being a member of the group set as the user's Monitor Group (User >
Telephony > Supervisor Settings). The user does not have to be a member of the group.

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 Call Listen

• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
A number of features are supported for call listening:
• Users can use privacy features to set a call cannot be intruded on and recorded.
• IP extensions can be monitored including those using direct media.
• The monitoring call can be initiated even if the target user is not currently on a call and
remains active until the monitoring user clears the monitoring call.
• The user who initiated the call listen can also record the call.
Intruding onto an a user doing silent monitoring (Call Listen) is turned into a silent monitoring call.
1400, 1600, 9500 and 9600 Series phones with a user button can initiate listening using that
button if the target user meets the criteria for listening.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Telephone Number: Target extension number (extension must be local).
• Default Short Code:
• Programmable Button Control: Listn
• See also: Call Intrude on page 960, Coaching Intrusion on page 976, Dial Inclusion on
page 984, Whisper Page on page 1040.
• Release: 1.0+.
Example
User 'Extn205' wants to be able to monitor calls received by members of the Hunt Group 'Sales'.
1. For user 'Extn205', in the Monitor Group (User > Telephony > Supervisor Settings) list
box, select the hunt group.
2. Ensure that Can Intrude is checked.
3. Create a user short code to allow Extn205 to start monitoring.
• Short Code: *89*N#
• Telephone Number: N
• Line Group ID: 0.
• Feature: CallListen
4. For each member of the hunt group, check that their Cannot be Intruded setting is
unchecked.
5. Now when a member of the 'Sales' hunt group is on a call, Extn205 can replace N in the
short code with the extension number of that member and monitor their call.
Related links
Short Code Features on page 953

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Short Code Features

Call Park
Parks the user's current call into the specified park slot number. The call can then be retrieved
by other extensions (refer to the appropriate telephone user guide). While parked the caller hears
music on hold if available. The 'Unpark Call' feature can be used to retrieve calls from specific
park slots.
Park Timeout (System | Telephony | Telephony) controls how long a call will remain parked. When
this expires the call will recall to the parking user if they are idle or when they next become idle.
The recall call will continue ring and does follow any forwards or go to voicemail.
Details
• Telephone Number: Park slot number.
- Park slot IDs can be up to 9 digits in length. Names can also be used for application park
slots.
- If no park slot number is specified when this short code is used, the system automatically
assigns a park slot number based on the extension number of the user parking the call
plus one digit 0 to 9.
• Default Short Code: *37*N#
• Programmable Button Control: Call Park
• See also: Unpark Call.
• Release: 1.0+.
Example
This short code is a default within the system configuration. This short code can be used to toggle
the feature on/off. N represents the park slot number in which the call will be parked. For example,
if a user wants to park a call to slot number 9, the user would dial *37*9#. The call will be parked
there until retrieved by another extension or the original extension.
• Short Code: *37*N#
• Telephone Number: N
• Feature: ParkCall
Related links
Short Code Features on page 953

Call Park and Page

Parks the user's current call into the highest park slot number within the range specified on the
System | Telephony | Park & Page tab, in the Central Park Range field. For example, if the
specified Central Park Range is 1XX, then the Park & Page short code would attempt to Park on
199. If the range is 567XX, then the call would attempt to Park on 56799.
Call Park and Page via short code is primarily useful for phones with no display, or phones on
which a Call Park operation is seldom performed. It provides a way for the user to Central Park

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 Call Pickup Any

in a pre-known location. If the highest Central Park slot is already in use, then the short code Call
Park and Page attempt will not be successful.
In order to perform a Page after a successful Call Park via short code, the user must enter a valid
Page short code.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: Call Park and Page
• Release: 9.0+.
Related links
Short Code Features on page 953

Call Pickup Any

Pick up the first available ringing call.
Details
• Telephone Number:
• Default Short Code: *30
• Programmable Button Control: PickA
• See also: Call Pickup Extn, Call Pickup Group, Call Pickup Members, Acquire Call, Call
Pickup Line, Call Pickup User.
• Release: 1.0+.
Example
Below is an example of the short code setup:
• Short Code: *30
• Feature: CallPickupAny
Related links
Short Code Features on page 953

Call Pickup Extn

Pick up a ringing call from a specific extension.

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Short Code Features

Details
• Telephone Number: Target extension number.
• Default Short Code: *32*N#
• Programmable Button Control: CpkUp
• See also: Call Pickup Any, Call Pickup Group, Call Pickup Members, Acquire Call, Call
Pickup Line, Call Pickup User.
• Release: 1.0+.
Example
This short code is a default within the system configuration. N represents the specific extension.
For example, if a user dials *32*201#, they will pick up the call coming into extension 201.
• Short Code: *32*N#
• Telephone Number: N
• Feature: CallPickupAny
Related links
Short Code Features on page 953

Call Pickup Group

Pick up a call ringing any hunt group of which the user is a member. The user can use this feature
even if their membership of the group is currently set as disabled.
Details
• Telephone Number:
• Default Short Code: *31
• Programmable Button Control: PickG
• See also: Call Pickup Any, Call Pickup Extn, Call Pickup Members, Acquire Call, Call Pickup
Line, Call Pickup User.
• Release: 1.0+.
Example
Below is an example of the short code setup.
• Short Code: *31
• Feature: CallPickupGroup
Related links
Short Code Features on page 953

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 Call Pickup Line

Call Pickup Line

Pick up an incoming call which is alerting, parked or held. The pickup uses the Line Appearance
ID specified in Telephone Number field of the short code. It cannot be used to pickup conferenced
calls. The normal user intrusion features are not applied to this pickup feature.
Details
• Telephone Number: Target Line Appearance ID.
• Default Short Code:
• Programmable Button Control:
• See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Call Pickup Members,
Acquire Call, Call Pickup User.
• Release: 4.0+ (Added in the Release 4.0 Q2 2007 maintenance release).
Example
This short code is a default within the system configuration. N represents the specific Line
Appearance ID.
• Short Code: *89*N#
• Telephone Number: N
• Feature: CallPickupLine
Related links
Short Code Features on page 953

Call Pickup Members

This feature can be used to pick up a ringing or queuing call at an extension that is a member of
the Hunt Group specified. The call picked up does not have to be a hunt group call. The function
includes group members even if their membership of the group is currently disabled.
Details
• Telephone Number: Group number or "Group name".
• Default Short Code: *53*N#
• Programmable Button Control: PickM
• See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Acquire Call, Call Pickup
Line, Call Pickup User.
• Release: 1.0+.

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Short Code Features

Example
Below is an example of the short code setup. N represents the extension number of the Hunt
Group. For example, if a user dials *53*500#, they will pick up the call coming into extension 500
(the hunt group's extension).
• Short Code: *53*N#
• Telephone Number: N
• Feature: CallPickupMembers
Related links
Short Code Features on page 953

Call Pickup User

Pick up an incoming call which is alerting, parked or held. The pickup uses the user extension
number specified in Telephone Number field of the short code. If there are multiple calls, priority is
given to picking up alerting, then parked and then held in that order of priority. It cannot be used to
pickup conferenced calls. The normal user intrusion features are not applied to this pickup feature.
Details
• Telephone Number: Target user extension number.
• Default Short Code:
• Programmable Button Control:
• See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Call Pickup Members,
Acquire Call, Call Pickup Line.
• Release: 4.0+.
Example
N represents the specific user.
• Short Code: *89*N#
• Telephone Number: N
• Feature: CallPickupUser
Related links
Short Code Features on page 953

Call Queue
Queue the current call to the destination phone, even when the destination phone is busy. This is
the same as a transfer except it allows you to transfer to a busy phone.

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 Call Record

Details
• Telephone Number: Target extension number.
• Default Short Code: *33*N#
• Programmable Button Control: Queue
• Release: 1.0+.
Example
Below is an example of the short code setup. N represents the extension the caller wishes to
queue for. For example, if a user dials *33*201# while connected to a caller, this caller will be
queued for extension 201.
• Short Code: *33*N#
• Telephone Number: N
• Feature: CallQueue
Related links
Short Code Features on page 953

Call Record
This feature allows you to record a conversation. To use this requires Voicemail Pro. Refer to your
local regulations in relation to the recording of calls.
• An advice of recording warning will be given if configured on the voicemail system.
• The recording is placed in the mailbox specified by the user's Manual Recording Mailbox
setting.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
• Users can use privacy features to set a call cannot be intruded on and recorded.
Details
• Telephone Number: Target extension number.
• Default Short Code:
• Programmable Button Control: Recor
• Release: 1.0+.
Example: Record your own extension's call
To use this short code, the user should place the call on hold and dial *55. They will automatically
be reconnected to the call when recording begins.
• Short Code: *55
• Telephone Number: None

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Short Code Features

• Feature: CallRecord
Related links
Short Code Features on page 953

Call Steal
This function allows a user to seize a call answered or ringing on another extension. This function
can be used with or without a specified user target.
• If the target has multiple alerting calls, the function steals the longest waiting call.
• If the target has a connected call and no altering calls, the function steals the connected call.
This is subject to the Can Intrude setting of the Call Steal user and the Cannot Be Intruded
setting of the target.
• If no target is specified, the function attempts to reclaim the user's last ringing or transferred
call if it has not been answered or gone to voicemail.
• Stealing a video call changes the call to an audio call.
• R11.1 FP2 SP4 and higher: The shortcode for this feature can be used with the user's
own extension number. That enables twinned and simultaneous device users to move a
connected call from another one of their devices. This usage ignores the user's privacy and
intrusion settings.
Details
• Telephone Number:
- Target extension number.
- User's own extension number to move call from other simultaneous device. This can
include using the U short code character.
- Blank for last call transferred.
• Default Short Code: *45*N# and *46
• Programmable Button Control: Aquire
• Release: 2.1+
Example: Taking Over a Call
In this example, N represents the extension to be taken over. For example, if a user dials
*45*201#, they will take over the current call on extension 201.
• Short Code: *45*N#
• Telephone Number: N
• Feature: Call Steal

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 Call Waiting On

Example: Reclaiming a Call

This short code reclaims the last call from your extension. This function is useful when you want to
catch a call you have just missed that has gone off to Voicemail.
• Short Code: *46
• Feature: Call Steal
Related links
Short Code Features on page 953

Call Waiting On
Enables call waiting on the user's extension. When on, if the user receives a second calls when
already on a call, they hear a call waiting tone in the speech path.
Call waiting settings are ignored for users with multiple call appearance buttons. In this case the
appearance buttons are used to indicate additional calls. Call waiting is automatically applied for
users with 'internal twinned' phones.
Details
• Telephone Number:
• Default Short Code: *15 (not on Server Edition)
• Programmable Button Control: CWOn
• See also: Call Waiting Off, Call Waiting Suspend.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *15
• Feature: CallWaitingOn
Related links
Short Code Features on page 953

Call Waiting Off

Disables call waiting on the user's extension. Call waiting may be applied for users with internal
twinned phones regardless of their call waiting settings.
Details
• Telephone Number:

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Short Code Features

• Default Short Code: *16 (not on Server Edition)

• Programmable Button Control: CWOff
• See also: Call Waiting On, Call Waiting Suspend.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *16
• Feature: Call Waiting Off
Related links
Short Code Features on page 953

Call Waiting Suspend

For phones using call waiting, this feature temporarily disables call waiting for the duration of the
user's next call.
Details
• Telephone Number:
• Default Short Code: *70 (A-Law only)
• Programmable Button Control: CWSus
• See also: Call Waiting On, Call Waiting Off.
• Release: 1.0+.
Example
Below is a sample of the short code setup. This short code is a default within the system
configuration.
• Short Code: *70
• Feature: CallWaitingSuspend
Related links
Short Code Features on page 953

Cancel All Forwarding

This feature cancels all forms of forwarding on the user's extension including "Follow Me" and "Do
Not Disturb".

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 Cancel Ring Back When Free

Details
• Telephone Number:
• Default Short Code: *00
• Programmable Button Control: FwdOf
• See also: Forward On Busy On, Forward On Busy Off, Forward On No Answer On, Forward
On No Answer Off, Forward Unconditional On, Forward Unconditional Off, Do Not Disturb
On, Do Not Disturb Off.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *00
• Feature: CancelCallForwarding
Related links
Short Code Features on page 953

Cancel Ring Back When Free

Cancels any existing ring back (also known as callback) set by the user.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: RBak-
• See also: Ring Back When Free.
• Release: 1.0+.
Example: Cancel Ring Back When Free
This example Short Code will cancel Ring Back When Free on the specified extension. N
represents the target extension from which you have set a ring back. For example, if Paul has
set a ring back on extension 201, he must dial *84*201# to cancel that ring back request.
• Short Code: *84*N#
• Telephone Number: N
• Feature: CancelRingBackWhenFree
Related links
Short Code Features on page 953

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Short Code Features

Change Login Code

Allows a user to change their login code. The login code must meet the Login Code Complexity
requirements defined on the System | Telephony tab.
Details
• Telephone Number: The user's current and new log in codes separated by a *, see the
examples below.
• Default Short Code:
• Programmable Button Control:
Example
The user has a Login Code of 1234 and wants to change it to 5678. To use the short code below,
the user must dial *60*1234*5678#.
• Short Code: *60*N#
• Telephone Number: N
• Feature: Change Login Code.
Example
For a user with no login code currently set, they can still use the short code to set a login code.
For example using the short code created above to set their login code to 1234 they should dial
*60**1234#.
Example
System phone users can also use this short code to change the login code of an other user. For
example 403 is configured as a system phone with a login code of 1234. User 410 has forgotten
their login code and needs it changed. User 403 can do this by dialing the following:
• *60*410*1234*<new code>#
Related links
Short Code Features on page 953

Clear After Call Work

This feature can be users who have been configured as CCR agents. It allows them to dial a short
code to exit the After Call Work (ACW) state as reported by the Customer Call Reporter (CCR)
application.
Note:
CCR is not supported in IP Office release 9.1 and later.
Details
• Telephone Number:

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 Clear Call

• Default Short Code:

• Programmable Button Control: ACWrk
• See also: Start After Call Work.
• Release: 4.2 4Q 2008 Maintenance release+.
Related links
Short Code Features on page 953

Clear Call
This feature can be used to end the current call.
Details
• Telephone Number:
• Default Short Code: *52
• Programmable Button Control: Clear
• Release: 1.0+.
Example
Below is a sample of the short code setup. This example could be used in a situation where you
are doing a supervised transfer and the party to be transferred to does not want to take the call. In
this scenario, you can put the call on hold and dial *52. This will clear the last connected call (for
example the party who has just refused the transfer), and retrieve the original call or dial tone.
• Short Code: *52
• Feature: Deny/ClearCall
Related links
Short Code Features on page 953

Clear CW
This feature is most commonly used to end the user's current call and answer the waiting call.
• Call waiting settings are ignored for users with multiple call appearance buttons.
Details
• Telephone Number:
• Default Short Code: *26 (A-Law only) (not on Server Edition)
• Programmable Button Control: ClrCW
• Release: 1.0+.

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Short Code Features

Example
Below is a sample of the short code setup.
• Short Code: *26
• Feature: ClearCW
Related links
Short Code Features on page 953

Clear Hunt Group Night Service

This feature changes the specified hunt group from Night Service mode to In Service mode.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is currently not supported between systems in a multi-site network. It can only be
used by a user currently logged onto the same system as hosting the hunt group.
Details
• Telephone Number:
- Hunt group extension number. If left blank, the short code will affect all hunt groups of
which the user is a member.
- The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively. The
service is indicated by setting the service name as the telephone number or action data.
Do not use quotation marks.
• Default Short Code: *21*N#
• Programmable Button Control: HGNS-
• See also: Clear Hunt Group Out Of Service, Set Hunt Group Night Service, Set Hunt Group
Out Of Service.
• Release: 1.0+.
Example
Below is a sample of the short code setup. N represents the telephone number of the hunt group
to be taken out of "Night Service" mode and placed into "In Service" mode. For example, when
*21*201# is dialed, the hunt group associated with extension 201 will be taken out of "Night
Service" mode.
• Short Code: *21*N#
• Telephone Number: N
• Feature: ClearHuntGroupNightService

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 Clear Hunt Group Out Of Service

Related links
Short Code Features on page 953

Clear Hunt Group Out Of Service

This feature changes the specified hunt group from Out of Service mode to In Service mode. This
will not override a hunt group in night service due to a time profile.
Details
• Telephone Number: Hunt group extension number. If left blank, the short code will affect
all hunt groups of which the user is a member.
• Default Short Code:
• Programmable Button Control: HGOS-
• See also: Clear Hunt Group Night Service, Set Hunt Group Night Service, Set Hunt Group
Out Of Service.
• Release: 1.0+.
Example
Below is a sample short code using the Clear Hunt Group Out Of Service feature. N represents
the telephone number of the hunt group to be taken out of "Out of Service" mode. For example,
when *55*201# is dialed, the hunt group associated with extension 201 will be placed into "In
Service" mode.
• Short Code: *55*N#
• Telephone Number: N
• Feature: ClearHuntGroupOutOfService
Related links
Short Code Features on page 953

Clear Quota
This feature refreshes the time quota for all services or a specific service.
Details
• Telephone Number: "Service name" or "" (all services).
• Default Short Code:
• Programmable Button Control: Quota
• Release: 1.0+.

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Short Code Features

Related links
Short Code Features on page 953

Coaching Intrusion
This feature allows the you to intrude on another user's call and to talk to them without being
heard by the other call parties to which they can still talk. For example: User A is on a call with
user B. When user C intrudes on user A, they can hear users A and B but can only be heard by
user A.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
• Listening to a call without the other parties being aware is subject to local regulations. You
must ensure that you have complied with the local regulations. Failure to do so can result in
penalties.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Telephone Number: Target extension number.
• Default Short Code:
• Programmable Button Control: Coach.
• See also: Call Intrude, Call Listen, Dial Inclusion, Whisper Page.
• Release:9.0+
Related links
Short Code Features on page 953

Conference Add
Conference add controls can be used to place the user, their current call and any calls they have
on hold into a conference. When used to start a new conference, the system automatically assigns
a conference ID to the call. This is termed ad-hoc (impromptu) conferencing.
If the call on hold is an existing conference, the user and any current call are added to that
conference. This can be used to add additional calls to an ad-hoc conference or to a meet-me
conference. Conference add can be used to connect two parties together. After creating the
conference, the user can drop from the conference and the two incoming calls remain connected.
For further details, see Conferencing on page 913.

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 Conference Meet Me

Details
• Telephone Number:
• Default Short Code: *47
• Programmable Button Control: Conf+
• See also: Conference Meet Me.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *47
• Feature: ConferenceAdd
Related links
Short Code Features on page 953

Conference Meet Me
Conference meet-me refers to features that allow a user or caller to join a specific conference by
using the conference's ID number (either pre-set in the control or entered at the time of joining the
conference).
Non-subscription IP500 V2 systems require a Preferred Edition license.
Note:
Conference Meet Me features can create conferences that include only one or two parties.
These are still conferences that are using resources from the host system's conference
capacity.
Conference ID Numbers
By default, ad hoc conferences are assigned numbers starting from 100 for the first conference
in progress. Therefore, for conference Meet Me features specify a number away from this range
ensure that the conference joined is not an ad hoc conference started by other users. It is no
longer possible to join a conference using conference Meet Me features when the conference ID is
in use by an ad-hoc conference.
User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system.

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Short Code Features

Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE 18 service.
Multi-Site Network Conferencing
Meet Me conference IDs are now shared across a multi-site network. For example, if a conference
with the ID 500 is started on one system, anyone else joining conference 500 on any system will
join the same conference. Each conference still uses the conference resources of the system on
which it was started and is limited by the available conference capacity of that system.
Previously separate conferences, each with the same conference ID, could be started on each
system in a multi-site network.
Other Features
Transfer to a Conference Button A currently connected caller can be transferred into the
conference by pressing TRANSFER, then the Conference Meet Me button and TRANSFER again
to complete the transfer. This allows the user to place callers into the conference specified by the
button without being part of the conference call themselves. This option is only support on Avaya
phones with a fixed TRANSFER button.
Conference Button Status Indication When the conference is active, any buttons associated
with the conference ID indicate the active state.
For further details, see Conferencing on page 913.
.
Details
• Telephone Number: Conference number. This can be an alphanumeric value up to 15
characters.
- The number can be prefixed with H(x) where x is the number of the music-on-hold source
that should be played to the first caller to enter the conference.
• Default Short Code: / *66*N# on Server Edition systems.
• Programmable Button Control: CnfMM
• See also: Conference Add.
• Release: 1.0+.
Related links
Short Code Features on page 953

CW
Pick up the waiting call. This feature provides same functionality as pressing the Recall or Hold
key on the phone. Unlike the Clear CW feature, this feature does not disconnect you from the
existing call when the second call is picked up.

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 Dial

Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial
This short code feature allows users to dial the number specified to an outside line.
Details
• Telephone Number: Telephone number.
• Default Short Code: Various depending on locale and system type.
• Programmable Button Control: Dial
• See also: Dial Direct, Dial Emergency, Dial Extn, Dial Inclusion, Dial Paging.
• Release: 1.0+.
Example: Creating a Speed Dial
In this example, users entering 401 on their telephone key pad will dial the New Jersey Office on
212 555 0000.
• Short Code: 401
• Telephone Number: 2125550000
Example: Replace Outgoing Caller ID
This short code is useful in a "call center" environment where you do not want customers to
have access to the number of your direct line; you want the general office number displayed. The
sample short code below will force the outgoing caller ID to display 123.
Use of this feature is dependent upon your local service provider.
• Short Code: ?
• Telephone Number: .s123
Example: External Dialing Prefix
The short code is for dialing a prefix for an outside line N represents the external number you want
to call.
• Short Code: 9N
• Telephone Number: N

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Short Code Features

Example: Blocking Caller ID

This is for blocking Caller ID for external calls. This feature can be applied to specific external
numbers or to all out going calls. In most situations, the company will choose to block the caller ID
for all external calls or leave it available for all external calls.
• Short Code: 9N
• Telephone Number: NW
Example: Maximum Call Length
The character t can be used in dialing short codes to set the maximum allowed duration of a
call. For example, the following short code will dial a number but then disconnect the call after 20
minutes (plus or minus a minute).
• Short Code: 9N
• Telephone Number: Nt(20)
Related links
Short Code Features on page 953

Dial 3K1
Sets the ISDN bearer capabilities to 3.1Khz audio call.
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: D3K1
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial 56K
Sets the ISDN bearer capabilities to 56Kbps data call.
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: D56K
• Release: 1.0+.

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 Dial 64K

Related links
Short Code Features on page 953

Dial 64K
Sets the ISDN bearer capabilities to 64Kbps data call.
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: D64K
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial CW
Call the specified extension number and force call waiting indication on if the extension is already
on a call.
If the user has call appearance buttons programmed, call waiting will not get activated. The next
incoming call will appear on an available call appearance button. When there are no available call
appearance buttons, the next incoming call will receive busy tone.
Details
• Telephone Number: Extension number.
• Default Short Code:
• Programmable Button Control: DCW
• Release: 1.0+.
Example
N represents the extension number to be dialed. For example, a user dialing *97*201# will force
call waiting indication on at extension 201 if extension 201 is already on a call.
• Short Code: *97*N#
• Telephone Number: N
• Feature: DialCW
Related links
Short Code Features on page 953

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Short Code Features

Dial Direct
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
Details
• Telephone Number: Extension number
• Default Short Code:
• Programmable Button Control: Dirct
• See also: Dial Paging.
• Release: 1.0+.
Example
This allows the extension specified to be automatically answered. N represents the extension that
will be forced to automatically answer. For example, when a user dials *83*201#, extension 201
will be forced to automatically answer the call.
• Short Code: *83*N#
• Telephone Number: N
• Feature: DialDirect
Related links
Short Code Features on page 953

Dial Direct Hot Line

When the line appearance button is mapped to a short code using the Dial Direct Hot Line short
code feature, no secondary dial tone is generated and the number is dialed directly. This feature
should not be confused with the hot line feature enabled using ?D short codes.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• Release: 3.0 to 4.0, 8.0+

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 Dial Emergency

Example
Below is a sample short code using the Dial Direct Hot Line feature. The short code *83* should
then be set as the prefix for the particular line required.
• Short Code: *83*
• Telephone Number: .
• Feature: DialDirectHotLine
Related links
Short Code Features on page 953

Dial Emergency
Dials the number specified regardless of any call barring applicable to the user.
On all systems, regardless of locale; system short codes using the Dial Emergency feature
should be created for any required emergency service numbers, with and without any external
dialing prefixes. Using a combination of location and emergency ARS entries, calls made
matching the emergency short codes should be routed to suitable lines. See Configuration for
Emergency Calls on page 652.
• Details of calls made using this function can be viewed using an Emergency View button.
See Emergency View on page 1103.
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: Emrgy
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial Extn
This feature can be used to dial an internal extension number (user or hunt group).
Details
• Telephone Number: Extension number.
- p( x ) can be added as a suffix to the Telephone Number to change the priority of a
call. Allowable values for x are 1, 2 or 3 for low, medium or high priority respectively. For
example Np(1).
• Default Short Code:

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Short Code Features

• Programmable Button Control:

• See also: Dial Direct, Dial Paging, DialPhysicalExtensionByNumber,
DialPhysicalNumberByID.
• Release: 1.0+.
Example: Dial on Pick up
The following user short code dials the extension specified the moment the user's handset is
picked up.
• Short Code: ?D
• Telephone Number: 201
• Line Group ID: 0
• Feature: Dial Extn
Related links
Short Code Features on page 953

Dial Fax
This feature is used to route fax calls via Fax Relay.
Details
• Telephone Number: Fax destination number.
• Default Short Code:
• Programmable Button Control:
• Release: 5.0+.
Example
In this example, the line group ID matches the URI configured on a SIP line that has been
configured for Fax Relay.
• Short Code: 6N
• Telephone Number: N"@192.16.42.5"
• Line Group ID: 17
• Feature: Dial Fax
Related links
Short Code Features on page 953

Dial Inclusion
This feature allows you to intrude on another user's call to talk to them. Their current call is put
on hold while you talk and automatically reconnected when you end the intrusion. The intruder

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 Dial Paging

and the target extension can then talk but cannot be heard by the other party. This can include
intruding into a conference call, where the conference will continue without the intrusion target.
During the intrusion all parties hear a repeated intrusion tone. When the intruder hangs-up the
original call parties are reconnected. Attempting to hold a dial inclusion call simply ends the
intrusion. The inclusion cannot be parked.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Release: 1.4+.
• See also: Call Intrude, Call Listen, Coaching Intrusion, Whisper Page.
• Programmable Button Control: Inclu.
• Default Short Code:
• Telephone Number: Target extension number.
Example
N represents the extension to be intruded upon. For example, if a user dials *97*201# while
extension 201 is on a call, then the user is intruding into extn. 201's current call.
• Short Code: *97*N#
• Telephone Number: N
• Feature: DialInclusion
Related links
Short Code Features on page 953

Dial Paging
This feature makes a page call to an extension or group. The target extension or group members
must support page calls (that is be able to auto-answer calls).
• When paging, always use only one codec (the preferred). It is the system administrator's
responsibility to ensure all the phones in the paging group support the codec.
Details
• Telephone Number: Extension or group number.
• Default Short Code:
• Programmable Button Control: Page

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Short Code Features

• See also: Dial Direct.

• Release: 1.0+.
Related links
Short Code Features on page 953

Dial Physical Extension by Number

Dial a specified extension number regardless of the current user logged in at that extension
and any forwarding, follow me or do not disturb settings applied by the current extension user.
Note that the extension number used is the Base Extension number set against the extension
configuration settings.
Details
• Telephone Number: Base Extension number.
• Default Short Code: *70*N# (U-Law only) (not on Server Edition)
• Programmable Button Control: PhyEx
• See also: Dial Physical Extension By Id, Priority Call.
• Release: 1.4+.
Example
The example below allows the extension with the base extension number 201 to be called
regardless of the extension number of the user currently logged in at that extension.
• Short Code: *97
• Telephone Number: 201
• Feature: DialPhysicalExtnByNumber
Related links
Short Code Features on page 953

Dial Physical Extension By ID

Dial a specific extension using its system ID. This may be necessary in hot desking environments
where some extensions have been created with no default extension number. Without an
extension number, a call can not be made to that extension unless a short code is created.
Details
• Telephone Number: Extension ID
• Default Short Code: *71*N# (U-Law only)

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 Dial Speech

• Programmable Button Control: DialP

• See also: DialPhysicalExtensionByNumber, Priority Call.
• Release: 1.4+.
Example
In the above example, if the telephone at extension ID 16 is not associated with an extension
number, a user can dial *97 to connect to that phone. This may be useful in hot desking
environments where some extensions may not have a dedicated base extension number.
• Short Code: *97
• Telephone Number: 16
• Feature: DialPhysicalNumberByID
Related links
Short Code Features on page 953

Dial Speech
This feature allows a short code to be created to force the outgoing call to use the Speech bearer
capability.
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: DSpch
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial V110
Sets the ISDN bearer capabilities to V110. The call is presented to local exchange as a "Data
Call".
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: DV110
• Release: 1.0+.

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Short Code Features

Related links
Short Code Features on page 953

Dial V120
Sets the ISDN bear capabilities using V.120.
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: DV120
• Release: 1.0+.
Related links
Short Code Features on page 953

Dial Video
The call is presented to the local exchange as a "Video Call".
Details
• Telephone Number: Telephone number.
• Default Short Code:
• Programmable Button Control: Dvide
• Release: 1.0+.
Related links
Short Code Features on page 953

Disable ARS Form

This feature can be used to put an ARS form out of service. It can be used with ARS forms for
which an Out of Service Route has been configured in Manager. The short code feature Enable
ARS Form can be used to return an ARS form to in service.
Details
• Telephone Number: ARS form number.

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 Disable Internal Forwards

• Default Short Code:

• Programmable Button Control:
• See also: Enable ARS Form
• Release: 4.0+.
Related links
Short Code Features on page 953

Disable Internal Forwards

This feature turns off the forwarding of internal calls for the user. It applies to Forward
Unconditional, Forward on Busy and Forward on No Answer.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Disable Internal Forwards Unconditional, Disable Internal Forwards Busy or
No Answer, Cancel All Forwarding, Enable Internal Forwards, Enable Internal Forwards
Unconditional, Enable Internal Forwards Busy or No Answer.
• Release: 3.2+.
Related links
Short Code Features on page 953

Disable Internal Forward Unconditional

This feature turns off the forwarding of internal calls for the user. It applies to Forward
Unconditional only.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Disable Internal Forwards, Disable Internal Forwards Busy or No Answer, Cancel
All Forwarding, Enable Internal Forwards, Enable Internal Forwards Unconditional, Enable
Internal Forwards Busy or No Answer.
• Release: 3.2+.

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Short Code Features

Related links
Short Code Features on page 953

Disable Internal Forward Busy or No Answer

This feature turns off the forwarding of internal calls for the user. It applies to Forward on Busy and
Forward on No Answer.
Details
• Telephone Number: No
• Default Short Code: No
• Programmable Button Control: No
•
• See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Cancel All
Forwarding, Enable Internal Forwards, Enable Internal Forwards Unconditional, Enable
Internal Forwards Busy or No Answer.
Related links
Short Code Features on page 953

Display Msg
Allows the sending of text messages to digital phones on the local system.
Details
• Telephone Number: The telephone number takes the format N";T" where:
- N is the target extension.
- T is the text message. Note that the "; before the text and the " after the text are
required.
• Default Short Code: No
• Programmable Button Control: Displ
Example
Below is a sample of the short code setup. When used, the target extension will hear a single ring
and then see the message. If the target extension is on a call then may need to scroll the display
to a free call appearance in order to see the text message.
• Telephone Number: N";Visitor in Reception"
• Feature: Display Msg

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 Do Not Disturb Exception Add

• Short Code: *78*N#

Example: SIP Extension Message Waiting Indicator
You can use the Display Msg short code to turn an extension’s message waiting indicator (MWI)
on or off.
• Telephone Number: The telephone number takes the format N";T" where:
- N is the target extension.
- T is the text message. Note that the "; before the text and the " after the text are
required.
• To turn MWI on, the telephone number must be N";Mailbox Msgs=1"
• To turn MWI off, the telephone number must be N";Mailbox Msgs=0"
• Default Short Code: No
Example
Below is a sample of the short code setup to turn MWI on. When used, the target extension will
receive a message directing it to turn the MWI on.
• Short Code: *99*N#
• Feature: Display Msg
• Telephone Number: N";Mailbox Msgs=1"
Example
Below is a sample of the short code setup to turn MWI off. When used, the target extension will
receive a message directing it to turn the MWI off.
• Short Code: *98*N#
• Feature: Display Msg
• Telephone Number: N";Mailbox Msgs=0"
Related links
Short Code Features on page 953

Do Not Disturb Exception Add

This feature adds a number to the user's "Do Not Disturb Exception Numbers List". This can be an
internal extension number or external ICLID. Calls from that number, except hunt group calls, will
ignore the user's Do Not Disturb setting. For further details see Do Not Disturb (DND).
Details
• Telephone Number: Telephone number or ICLID. Up to 31 characters. For ICLID numbers
any prefix added by the system must also be included.
• Default Short Code: *10*N#

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Short Code Features

• Programmable Button Control: DNDX+

• See also: Do Not Disturb Exception Delete, Do Not Disturb On, Do Not Disturb Off.
Example
N represents the number to be added to the user's "Do Not Disturb Exception List". For example,
when a user has DND turned on and dials *10*4085551234#, incoming calls from telephone
number (408) 555-1234. All other calls, except those numbers on the exception list hear busy
tones or are redirected to voicemail if available.
• Short Code: *10*N#
• Telephone Number: N
• Feature: DoNotDisturbExceptionAdd
Example
In this example, the last number received by the user is added to their exception list.
• Short Code: *89
• Telephone Number: L
• Feature: DoNotDisturbExceptionAdd
Related links
Short Code Features on page 953

Do Not Disturb Exception Delete

This feature removes a number from the user's "Do Not Disturb Exception List". For further details
see Do Not Disturb (DND).
Details
• Telephone Number: Telephone number or ICLID.
• Default Short Code: *11*N#
• Programmable Button Control: DNDX-
• See also: Do Not Disturb Exception Add, Do Not Disturb On, Do Not Disturb Off.
• Release: 1.0+.
Example
N represents the number to be deleted from the user's "Do Not Disturb Exception List". For
example, when a user has DND turned on and the telephone number (408) 555-1234 in their "Do
Not Disturb Exception List", dialing *10*4085551234# will remove this phone number from the list.
Incoming calls from (408) 555-1234 will no longer be allowed through; instead they will hear busy
tone or be redirected to voicemail if available.
• Short Code: *11*N#
• Telephone Number: N

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 Do Not Disturb On

• Feature: DoNotDisturbExceptionDel
Related links
Short Code Features on page 953

Do Not Disturb On
This feature puts the user into 'Do Not Disturb' mode. When on, all calls, except those from
numbers in the user's exception list hear busy tones or are redirected to voicemail if available. For
further details, see Do Not Disturb (DND).
• CCR is not supported in IP Office release 9.1 and later.
Details
• Telephone Number:
• Default Short Code: *08
• Programmable Button Control: DNDOn
• See also: Do Not Disturb Off, Do Not Disturb Exception Add, Do Not Disturb Exception
Delete.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *08
• Feature: DoNotDisturbOn
Related links
Short Code Features on page 953

Do Not Disturb Off

Cancels the user's 'do not disturb' mode if set. For further details, see Do Not Disturb (DND).
Details
• Telephone Number:
• Default Short Code: *09
• Programmable Button Control: DNDOf
• See also: Do Not Disturb On, Do Not Disturb Exception Add, Do Not Disturb Exception
Delete.
• Release: 1.0+.

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Short Code Features

Example
This short code is a default within the system configuration. Below is a sample of the short code
setup.
• Short Code: *09
• Feature: DoNotDisturbOff
Related links
Short Code Features on page 953

Enable ARS Form

This feature can be used to put an ARS form in service. It can be used with ARS forms that have
been put out of service through Manager or the use of a Disable ARS Form short code.
Details
• Telephone Number: ARS form number.
• Default Short Code:
• Programmable Button Control:
• Release: 4.0+
Related links
Short Code Features on page 953

Enable Internal Forwards

This feature turns on the forwarding of internal calls for the user. It applies to Forward
Unconditional, Forward on Busy and Forward on No Answer.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable
Internal Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards
Unconditional, Enable Internal Forwards Busy or No Answer.
• Release: 3.2+.
Related links
Short Code Features on page 953

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 Enable Internal Forward Unconditional

Enable Internal Forward Unconditional

This feature turns on the forwarding of internal calls for the user. It applies to Forward
Unconditional only.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable
Internal Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards,
Enable Internal Forwards Busy or No Answer.
• Release: 3.2+.
Related links
Short Code Features on page 953

Enable Internal Forward Busy or No Answer

This feature turns on the forwarding of internal calls for the user. It applies to Forward on Busy and
Forward on No Answer.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable
Internal Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards,
Enable Internal Forwards Unconditional.
• Release: 3.2+.
Related links
Short Code Features on page 953

Extn Login
Extn Login allows a user who has been configured with a Login Code (User | Telephony |
Supervisor Settings) to take over ownership of any extension. That user's extension number

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Short Code Features

becomes the extension number of the extension while they are logged. This is also known as ‘hot
desking’.
• Hot desking is not supported for H175 and J129 telephones.
• When used, the user will be prompted to enter their extension number and then their log in
code. Login codes of up to 15 digits are supported with Extn Login buttons. Login codes of
up to 31 digits are supported with Extn Login short codes.
• When a user logs in, as many of their user settings as possible are applied to the extension.
The range of settings applied depends on the phone type and on the system configuration.
• By default, on 1400 Series, 1600 Series, 9500 Series and 9600 Series phones, the user's
call log and personal directory are accessible while they are logged in. This also applied to
M-Series and T-Series telephones.
• On other types of phone, those items such as call logs and speed dials are typically stored
locally by the phone and will not change when users log in and log out.
• If the user logging in was already logged in or associated with another phone, they will be
automatically logged out that phone.
Details
• Telephone Number: Extension Number*Login Code. If just a single number is dialed
containing no * separator, the system assumes that the extension number to use is the
physical extension's Base Extension number and that the number dialed is the log in code.
• Default Short Code: *35*N#
• Programmable Button Control: Login
• See also: Extn Logout.
• Release: 1.0+.
Example: Individual Hot Desking
Based on the above sample short code, Paul (extension 204) can go to another phone (even if it
is already logged in by another user) and log in as extension 204 by simply dialing 299. Once Paul
has logged into this phone, extension 204 is logged out at Paul's original phone. For Paul to make
use of this short code, his log in code must match that configured in the above short code. When
Paul logs out of the phone he has "borrowed", his original extension will automatically be logged
back in.
• Short Code: 299
• Telephone Number: 204*1234
• Feature: Extnlogin
Example: Log In
The default short code for logging into a phone is configured as shown below. N represents the
users extension number followed by a * and then their log in code, for example *35*401*123#.
• Short Code: *35*N#
• Telephone: N

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 Extn Logout

• Feature: ExtnLogin
Related links
Short Code Features on page 953

Extn Logout
This feature logs the user off the phone at which they are logged in. This feature cannot be used
by a user who does not have a log in code or by the default associated user of an extension
unless they are set to forced log in.
Details
• Telephone Number:
• Default Short Code: *36
• Programmable Button Control: Logof
• See also: Extn Login.
• Release: 1.0+.
Example
Below is a sample short code using the Extn Logout feature. This short code is a default within the
system configuration.
• Short Code: *36
• Feature: ExtnLogout
Related links
Short Code Features on page 953

Flash Hook
This feature sends a hook flash signal to the currently connected line if it is an analog line.
Only supported for analog lines on the same system as the short code. See Centrex Transfer on
page 791.
Details
• Telephone Number: Optional The telephone number field can be used to set the transfer
destination number for a Centrex Transfer. In this case the use of the short code Forced
Account Code and Forced Authorization Code are not supported and the Line Group Id must
match the outgoing line to the Centrex service provider.
• Default Short Code:
• Programmable Button Control: Flash

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Short Code Features

• Release: 1.4+.
Example
Below is a sample short code using the Flash Hook feature.
• Short Code: *96
• Feature: FlashHook
Related links
Short Code Features on page 953

FNE Service
This short code feature is used for Mobile Call Control and one-X Mobile Client support.
Details
• Telephone Number: This number sets the required FNE function.
• Default Short Code:
• Programmable Button Control:
• Release: 4.2+.
Related links
Short Code Features on page 953

Follow Me Here
Causes calls to the extension number specified to be redirected to the extension initiating the
'Follow Me Here'. If the redirected call receives a busy tone or is not answered, then the call
behaves as though the User's extension had failed to answer. For further details, see Follow
Me on page 747.
Details
Telephone Number: Extension to redirect to the dialing extension.
Default Short Code: *12*N#
Programmable Button Control: Here+
See also: Follow Me Here Cancel, Follow Me To.
Release: 1.0+.
Example
This feature is used at the Follow Me destination. N represents the extension number of the user
wanting their calls redirected to that destination. For example, User A's extension is 224. However

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 Follow Me Here Cancel

they are working at extension 201 and want their calls redirected there. If the following short code
is available, they can do this by dialing *12*224# at extension 201.
• Short Code: *12*N#
• Telephone Number: N
• Feature: FollowMeHere
Related links
Short Code Features on page 953

Follow Me Here Cancel

Cancels any Follow Me set on the specified extension. This action can only be performed at
the extension to which the Follow Me Here is targetted. For further details, see Follow Me on
page 747.
Details
• Telephone Number: Extension being redirected to the dialing extension.
• Default Short Code: *13*N#
• Programmable Button Control: Here-
• See also: Follow Me Here, Follow Me To.
• Release: 1.0+.
Example
This feature is used at the Follow Me destination. N represents the extension number of the user
whose calls are being redirected to that destination. For example, User A's extension is 224.
However they are working at extension 201 and so have set a Follow Me on their own extension
to redirect their calls to 201. If the following short code is available, they can cancel the Follow Me
by dialing *13*224# at extension 201.
Short Code: *13*N#
Telephone Number: N
Feature: FollowMeHereCancel
Related links
Short Code Features on page 953

Follow Me To
Causes calls to the extension to be redirected to the Follow Me destination extension specified.
For further details, see Follow Me on page 747.

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Short Code Features

Details
• Telephone Number: Target extension number or blank (cancel Follow Me To)
• Default Short Code: *14*N#
• Programmable Button Control: FolTo
• See also: Follow Me Here, Follow Me Here Cancel.
• Release: 1.0+.
Example
This feature is used at the extension that wants to be redirected. N represents the extension
number to which the user wants their calls redirected. For example, User A's extension is 224.
However they are working at extension 201 and want their calls redirected there. If the following
short code is available, they can do this by dialing *14*201# at extension 224.
• Short Code: *14*N#
• Telephone Number: N
• Feature: FollowMeTo
Related links
Short Code Features on page 953

Forward Hunt Group Calls On

Forward the user's hunt group calls (internal and external) to their forward number when the user
has Forward Unconditional active. For further details see Forward Unconditional.
This option is only applied for calls to Sequential and Rotary type hunt groups. Calls from other
types of hunt group types are not presented to the user when they have Forward Unconditional
active. Note also that hunt group calls cannot be forwarded to another hunt group.
Details
• Telephone Number:
• Default Short Code: *50
• Programmable Button Control: FwdH+
• See also: Forward Hunt Group Calls Off, Forward Unconditional On, Forward Unconditional
Off.
• Release: 1.0+.
Example
This short code is useful if the hunt group member temporarily uses another workstation and so
does not require a permanent extension change.
• Short Code: *50
• Feature: ForwardHuntgroupCallsOn

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 Forward Hunt Group Calls Off

Related links
Short Code Features on page 953

Forward Hunt Group Calls Off

This feature cancels the forwarding of the user's hunt group calls. For further details see Forward
Unconditional.
Details
• Telephone Number:
• Default Short Code: *51
• Programmable Button Control: FwdH-
• See also: Forward Hunt Group Calls On, Forward Unconditional On, Forward Unconditional
Off.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *51
• Feature: ForwardHuntgroupCallsOff
Related links
Short Code Features on page 953

Forward Number
Sets the number to which the user's calls are redirected. This can be an internal or external
number. The number is still subject to the user's call barring settings. For further details see
Forward Unconditional.
This feature does not activate forwarding; it only sets the number for the forwarding destination.
This number is used for all forward types; Forward Unconditional, Forward on Busy and Forward
on No Answer, unless the user has a separate Forward on Busy Number set for forward on busy
and forward on no answer functions.
Details
• Telephone Number: Telephone number.
• Default Short Code: *07*N#
• Programmable Button Control: FwdNo

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Short Code Features

• See also: Forward On Busy Number.

• Release: 1.0+.
Example
N represents the forward destination. For example, if extension 224 wants to set the forwarding
number to extension 201, the user can dial *07*201#.
• Short Code: *07N*#
• Telephone Number: N
• Feature: ForwardNumber
Related links
Short Code Features on page 953

Forward On Busy Number

Sets the number to which the user's calls are forwarded when Forward on Busy or Forward on No
Answer are on. If no Forward on Busy Number is set, those functions use the Forward Number.
For further details, see Forward on Busy on page 751.
This feature does not activate the forwarding, it only sets the number for the forwarding
destination.
Details
• Telephone Number: Telephone number.
• Default Short Code: *57*N#
• Programmable Button Control: FwBNo
• See also: Forward Number.
• Release: 1.0+.
Example
N represents the extension number to be forwarded to. For example, if Paul (whose extension is
224) wants to set the forwarding number for his 'Forward on Busy' and/or 'Forward on No Answer'
feature to extension 201, Paul can dial *57*201# followed by the short code for the forwarding
function.
• Short Code: *57N*#
• Telephone Number: N
• Feature: ForwardOnBusyNumber
Related links
Short Code Features on page 953

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 Forward On Busy On

Forward On Busy On
This feature enables forwarding when the user's extension is busy. It uses the Forward Number
destination or, if set, the Forward on Busy Number destination. If the user has call appearance
buttons programmed, the system will not treat them as busy until all the call appearance buttons
are in use. For further details, see Forward on Busy on page 751.
Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Details
• Telephone Number:
• Default Short Code: *03
• Programmable Button Control: FwBOn
• See also: Forward On Busy Off, Cancel All Forwarding, Enable Internal Forward Busy or No
Answer.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *03
• Feature: ForwardOnBusyOn
Related links
Short Code Features on page 953

Forward On Busy Off

This feature cancels forwarding when the user's extension is busy.
Details
• Telephone Number:
• Default Short Code: *04
• Programmable Button Control: FwBOf
• See also: Forward On Busy On, Cancel All Forwarding.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *04
• Feature: ForwardOnBusyOff

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Short Code Features

Related links
Short Code Features on page 953

Forward On No Answer On
This feature enables forwarding when the user's extension is not answered within the period
defined by their No Answer Time. It uses the Forward Number destination or, if set, the Forward
on Busy Number destination. For further details, see Forward on No Answer on page 753.
Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Details
• Telephone Number:
• Default Short Code: *05
• Programmable Button Control: FwNOn
• See also: Forward On No Answer Off, Cancel All Forwarding.
• Release: 1.0+.
Example
Below is a sample of the short code setup. Remember that the forwarding number for this feature
uses the 'Forward on Busy Number'.
• Short Code: *05
• Feature: ForwardOnNoAnswerOn
Related links
Short Code Features on page 953

Forward On No Answer Off

This feature cancels forwarding when the user's extension is not answered.
Details
• Telephone Number:
• Default Short Code: *06
• Programmable Button Control: FwNOf
• See also: Forward On No Answer On.
• Release: 1.0+.

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 Forward Unconditional On

Example
Below is a sample of the short code setup.
• Short Code: *06
• Feature: ForwardOnNoAnswerOff
Related links
Short Code Features on page 953

Forward Unconditional On
This feature enables forwarding of all calls, except group calls, to the Forward Number set for the
user's extension. To also forward hunt group calls, Forward Hunt Group Calls On must also be
used. For further details, see Forward Unconditional on page 749.
Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: FwUOn
• See also: Forward Unconditional Off.
• Release: 1.0+.
Example
Remember that this feature requires having a forward number configured.
• Short Code: *01
• Feature: ForwardUnconditionalOn
Related links
Short Code Features on page 953

Forward Unconditional Off

This feature cancels forwarding of all calls from the user's extension.
• This does not disable Forward on No Answer and or Forward on Busy if those functions are
also on. For further details see Forward Unconditional.

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Short Code Features

Details
• Telephone Number:
• Default Short Code: *02
• Programmable Button Control: FwUOf
• See also: Forward Unconditional On.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *02
• Feature: ForwardUnconditionalOff
Related links
Short Code Features on page 953

Group Listen Off

Disables the group listen function on the user’s extension. See Group Listen On on page 1006.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: GroupListenOn
• Release: 4.1+.
Related links
Short Code Features on page 953

Group Listen On
Using group listen allows callers to be heard through the phone's handsfree speaker but to only
hear the phone's handset microphone. When group listen is enabled, it modifies the handsfree
functionality of the user’s phone in the following manner
• When the user’s phone is placed in handsfree/speaker mode, the speech path from the
connected party is broadcast on the phone speaker but the phone's base microphone is
disabled.
• The connected party can only hear speech delivered through the phone's handset
microphone.

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 Headset Toggle

• Group listen is not supported for IP phones or when using a phone's HEADSET button.
• For T-Series and M- Series phones, this option can be turned on or off during a call. For other
phones, currently connected calls are not affected by changes to this setting, instead group
listen must be selected before the call is connected.
Group listen is automatically turned off when the call is ended.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: GroupListenOn
• Release: 4.1+.
Related links
Short Code Features on page 953

Headset Toggle
Toggles between the use of a headset and the telephone handset.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: HdSet
• Release: 1.4+.
Example
Below is a sample short code using the Headset Toggle feature. This short code can be used to
toggle the feature on/off. If an Avaya supported headset is connected to your telephone, this short
code can be used to toggle between using the headset and the telephone handset.
• Short Code: *55
• Feature: HeadsetToggle
Related links
Short Code Features on page 953

Hold Call
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The Hold Call feature "holds" the current call to a slot. The current call is

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Short Code Features

always automatically placed into slot 0 if it has not been placed in a specified slot. Only available if
supported by the ISDN exchange.
Details
• Telephone Number: Exchange hold slot number or blank (slot 0).
• Default Short Code:
• Programmable Button Control: Hold
• See also: Hold CW, Hold Music, Suspend Call.
• Release: 1.0+.
Example
Below is a sample short code using the Hold Call feature. This short code is a default within the
system configuration. N represents the exchange hold slot number you want to hold the call on.
For example, while connected to a call, dialing *24*3# will hold the call onto slot 3 on the ISDN.
• Short Code: *24*N#
• Telephone Number: N
• Feature: HoldCall
Related links
Short Code Features on page 953

Hold CW
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing
up the ISDN B channel. The Hold CW feature "holds" the current call to an exchange slot and
answers the call waiting. The current call is always automatically placed into slot 0 if it has not
been placed in a specified slot. Only available if supported by the ISDN exchange.
Details
• Telephone Number: Exchange slot number or blank (slot 0).
• Default Short Code: *27*N# (A-Law only) (not on Server Edition)
• Programmable Button Control: HoldCW
• See also: Hold Call, Suspend Call.
• Release: 1.0+.
Example
Below is a sample short code using the Hold CW feature.
• Short Code: *27*N#
• Feature: HoldCW
Related links
Short Code Features on page 953

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 Hold Music

Hold Music
This feature allows the user to check the system's music on hold. See Music On Hold for more
information.
Details
• Telephone Number: Optional. If no number is specified, the default system source is
assumed. The system supports up to 4 hold music sources, numbered 1 to 4. 1 represents
the System Source. 2 to 4 represent the Alternate Sources.
• Default Short Code:
• *34N; where N is the number of the hold music source required.
• Programmable Button Control: Music
• Release: 1.0+.
Example
Below is a sample short code using the Hold Music feature. This short code is a default within the
configuration.
• Short Code: *34N;
• Feature: HoldMusic
Related links
Short Code Features on page 953

Hunt Group Disable

This feature disables the user's membership of the specified hunt group. They will no longer
receive call to that hunt group until their membership is enabled again. To use this feature, you
must already belong to the hunt group. See also Hunt Group Enable.
Details
• Telephone Number: Group number.
• Default Short Code:
• Programmable Button Control: HGDis
• See also: Hunt Group Enable.
• Release: 1.0+.
Example
N represents the hunt group number from which the user wants to be disabled from. For example,
if Paul wants to be disabled from the Sales hunt group (extn. 500), he needs to dial *90*500#.
• Short Code: *90*N#
• Telephone Number: N

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Short Code Features

• Feature: HuntGroupDisable
Related links
Short Code Features on page 953

Hunt Group Enable

This feature enables the user's membership of a hunt group so they can begin to receive calls to
the specified hunt group. To use this feature, the user must already belong to the hunt group. This
short code can not be used to add someone to a hunt group, that must be done within Manager's
Hunt Group form.
Details
• Telephone Number: Group number.
• Default Short Code:
• Programmable Button Control: HGEna
• See also: Hunt Group Disable.
• Release: 1.0+. Previously in Release 3.2 the Set Hunt Group Night Service, Set Hunt
Group Out of Service and Hunt Group Enable short code features toggled. That behaviour
is not supported in 4.0 and higher.
Example
This short code can be used to turn the feature on. N represents the hunt group number for which
the user wants to start receiving calls. For example, if Paul is already a member of the sales
hunt group (extn. 500) but has changed his availability status for that hunt group using hunt group
disable, he can make himself available for receiving calls to the Sales hunt group again by dialing
*91*500#.
• Short Code: *91*N#
• Telephone Number: N
• Feature: HuntGroupEnable
Related links
Short Code Features on page 953

Last Number Redial

This feature allows an extension to redial the last number they dialed.
Details
• Telephone Number:

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 MCID Activate

• Default Short Code:

• Programmable Button Control:
• Release: 3.0+.
Related links
Short Code Features on page 953

MCID Activate
This feature should only be used in agreement with the ISDN service provider and the appropriate
local legal authorities. It allows users with Can Trace Calls (User | Telephony | Supervisor
Settings) set to trigger a malicious call trace of their previous call at the ISDN exchange. Refer to
Telephone Features Malicious Call Tracing for further details.
• Currently, in Server Edition network, MCID is only supported for users using an MCID button
and registered on the same IP500 V2 Expansion system as the MCID trunks.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: Advanced | Miscellaneous | MCID Activate.
• Release: 4.0+.
Related links
Short Code Features on page 953

Mobile Twinned Call Pickup

This short code feature allows the user to pickup a call ringing or connected at the destination of
their mobile twinning number. This short code can only be used from the primary extension which
is being used for the twinning operation.
Note that the use of mobile twinning requires entry of a Mobile Twinning license and may be
subject to a time profile.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Set Mobile Twinning Number, Set Mobile Twinning On, Set Mobile Twinning Off.

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Short Code Features

• Release: 3.2+.
Related links
Short Code Features on page 953

Off Hook Station

Enables or disables whether the user's extension acts as a fully handsfree unit. Typically this is
used when the answering and clearing of calls is done through an application. For more details
see Off Hook Station (User | Telephony | Call Settings).
Details
• Telephone Number: "Y" for on or "N" for off.
• Default Short Code:
• Programmable Button Control: OHStn
• Release: 1.0+.
Example: Turning the off hook station off
• Short Code: *89
• Telephone Number: N
• Feature: OffHookStation
Example: Turning the off hook station on
• Short Code: *98
• Telephone Number: Y
• Feature: OffHookStation
Related links
Short Code Features on page 953

Outgoing Call Bar Off

Allows a user to switch off their outgoing call bar status. The short code user must enter their log
in code, if set, in order to be successful.
If you add a short code using this feature to a system it is recommended that you also assign a
login code to the No User user to prevent the short code being used to change the status of that
user.

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 Outgoing Call Bar On

Details
• Telephone Number: The user's log in code.
- System phone users can use <target user>*<system phone user's login code>.
• Default Short Code:
• Programmable Button Control:
• Release: 4.1+ (Added to Release 4.1 2008Q2 Maintenance release).
Example
The user has a Login Code of 1234. To use the short code below, the user must dial *59*1234#.
• Short Code: *59*N#
• Telephone Number: N
• Feature: Outgoing Call Bar Off.
Example
A user set as a system phone can also switch off the Outgoing Call Bar status of another user.
This is done using their own login code. For example the system phone 401 with login code 1234
can switch off the outgoing call bar status of extension 403 as follows:
• *59*403*1234
Related links
Short Code Features on page 953

Outgoing Call Bar On

Allows a user to switch on their outgoing call bar status.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• Release: 4.1+ (Added to Release 4.1 2008Q2 Maintenance release).
Example
To use the short code below, the user must dial *58.
• Short Code: *58
• Telephone Number: <blank>
• Feature: Outgoing Call Bar On.
Related links
Short Code Features on page 953

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Short Code Features

Private Call Off

Short codes using this feature turn off private call status for the user if set. The short code features
Private Call and Private Call On can be used to turn private call on.
• When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private
call.
• Note that use of private calls is separate from the user's intrusion settings. If the user's
Cannot be Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching
private calls off does not affect that status. To allow private calls to be used to fully control the
user status, Cannot be Intruded (User | Telephony | Supervisor Settings) should be disabled
for the user.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: Advanced | Call | Private Call.
• Release: 4.0+.
Related links
Short Code Features on page 953

Private Call On
Short codes using this feature turn on the private call settings for the user regardless.
• When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private
call.
• Note that use of private calls is separate from the user's intrusion settings. If the user's
Cannot be Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching
private calls off does not affect that status. To allow private calls to be used to fully control the
user status, Cannot be Intruded (User | Telephony | Supervisor Settings) should be disabled
for the user.
• Private call status can be switched off using a short code with the Private Call Off feature or
a programmed button set to the Private Call action. To enable private call status for a single
following call only the Private Call short code feature should be used.
Details
• Telephone Number:
• Default Short Code:

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 Priority Call

• Programmable Button Control: Advanced | Call | Private Call.

• Release: 4.0+.
Related links
Short Code Features on page 953

Priority Call
This feature allows the user to call another user even if they are set to 'do not disturb'. Priority calls
to a user without DND will follow forwarding and follow me settings but will not go to voicemail.
Details
• Telephone Number: Extension number.
• Default Short Code:
• Programmable Button Control: PCall
• See also: DialPhysicalExtensionByNumber, DialPhysicalNumberByID.
• Release: 1.0+.
Example
N represents the extension number to be called, despite the extension being set to 'do not disturb'.
For example, if extension 201 has 'do not disturb' enabled, a user can dial *71*201# and still get
through. This short code is useful for companies that frequently use the 'do not disturb' feature and
can be given to Managing Directors or people who may need to get through to people regardless
of their 'do not disturb' status.
• Short Code: *71*N#
• Telephone Number: N
• Feature: PriorityCall
Related links
Short Code Features on page 953

Record Message
This short code feature is used to record hunt group announcements on Embedded Voicemail,
see Hunt Group | Announcements. Release 5.0+: It is also used to record mailbox user name
prompts for the auto attendant Dial by Name function.
Details
• Telephone Number:
- For a hunt group queue announcement, use the hunt group extension number followed by
".1".

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Short Code Features

- For a hunt group still queue announcement, use the hunt group extension number followed
by ".2".
- For a mailbox user name prompt, use the user extension number followed by ".3".
• Default Short Code: *91N; and *92N; (not on Server Edition)
• Programmable Button Control:
• Release: 4.0+.
Example
For a hunt group with extension number 300, the default short codes *91N;/Record Message/
N".1" and *92N;/Record Message/N".2" can be used to allow recording of the announcements
by dialing *91300# and *92300#.
To allow users to record their own name prompt, the short code *89#/Record Message/E."3" can
be used. The E is replace by the extension number of the dialing user.
Related links
Short Code Features on page 953

Relay On
This feature closes the specified switch in the system's external output (EXT O/P) port.
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Details
• Telephone Number: Switch number (1 or 2).
• Default Short Code: *39 (Switch 1), *42 (Switch 2), *9000*.
• Programmable Button Control: Rely+
• See also: Relay Off, Relay Pulse.
• Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based
on this sample short code, a user dialing *42 is closing switch number 2 to activate an external
device.
• Short Code: *42
• Telephone Number: 2
• Feature: RelayOn
Analog Modem Control
On systems with an analog trunk card in the control unit, the first analog trunk can be set to
answer V.32 modem calls. This is done by either selecting the Modem Enabled option on the

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 Relay Off

analog line settings or using the default short code *9000* to toggle this service on or off. This
short code uses the RelayOn feature with the Telephone Number set to "MAINTENANCE". Note
that the short code method is always returned to off following a reboot or if used for accessing the
system date and time menu.
IP500 ATM4 Uni Trunk Card Modem Support It is not required to switch the card's modem port
on/off. The trunk card's V32 modem function can be accessed simply by routing a modem call to
the RAS service's extension number. The modem call does not have to use the first analog trunk,
instead the port remains available for voice calls.
Related links
Short Code Features on page 953

Relay Off
This feature opens the specified switch in the system's external output (EXT O/P) port.
Details
• Telephone Number: Switch number (1 or 2).
• Default Short Code: *40 (Switch 1), *43 (Switch 2)
• Programmable Button Control: Rely-
• See also: Relay On, Relay Pulse.
• Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based
on this sample short code, a user dialing *43 is opening switch number 2 to activate an external
device.
• Short Code: *43
• Telephone Number: 2
• Feature: RelayOff
Related links
Short Code Features on page 953

Relay Pulse
This feature closes the specified switch in the system's external output (EXT O/P) port for 5
seconds and then opens the switch.

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Short Code Features

Details
• Telephone Number: Switch number (1 or 2).
• Default Short Code: *41 (Switch 1), *44 (Switch 2)
• Programmable Button Control: Relay
• See also: Relay On, Relay Off.
• Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based
on this sample short code, a user dialing *44 is opening switch number 2 to activate an external
device.
• Short Code: *44
• Telephone Number: 2
• Feature: RelayPulse
Related links
Short Code Features on page 953

Resume Call
Resume a call previously suspended to the specified ISDN exchange slot. The suspended call
may be resumed from another phone/ISDN Control Unit on the same line.
Details
• Telephone Number: Exchange suspend slot number.
• Default Short Code: *23*N# (A-Law only) (not on Server Edition)
• Programmable Button Control: Resum
• See also: Suspend Call.
• Release: 1.0+.
Example
Below is sample short code using the Resume Call feature. N represents the exchange slot
number from which the call has been suspended. For example, if a user has suspended a call on
slot number 4, this user can resume that call by dialing *23*4#.
• Short Code: *23*N#
• Telephone Number: N
• Feature: ResumeCall
Related links
Short Code Features on page 953

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 Retrieve Call

Retrieve Call
Retrieves a call previously held to a specific ISDN exchange slot.
Details
• Telephone Number: Exchange hold slot number.
• Default Short Code: *25*N# (A-Law only) (not on Server Edition)
• Programmable Button Control: Retriv
• See also: Hold Call.
• Release: 1.0+.
Example
Below is sample short code using the Retrieve Call feature. N represents the exchange slot
number from which the call has been placed on hold. For example, if a user has placed a call hold
on slot number 4, the user can resume that call by dialing *25*4#.
• Short Code: *25*N#
• Telephone Number: N
• Feature: RetrieveCall
Related links
Short Code Features on page 953

Ring Back When Free

This feature sets a ringback on the specified extension. This sets a 'ringback when free' on an
extension currently on a call or a 'ringback when next used' for an extension that is free but does
not answer.
When the target extension is next used or ends its current call, the users is rung and when they
answer a call is made to the target extension.
Details
• Telephone Number: Target extension number.
• Default Short Code:
• Programmable Button Control: RBak+
• See also: Cancel Ring Back When Free.
• Release: 1.0+.
Example
N represents the target extension from which you want to receive the callback. For example, if
you call extension 201 but the line is busy, hang up and then dial *71*201#. When extension 201

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Short Code Features

disconnects from its current call, your phone will ring. Once you pick up the phone, extension
201's line will start ringing to indicate an incoming call.
• Short Code: *71*N#
• Telephone Number: N
• Feature: RingBackWhenFree
Related links
Short Code Features on page 953

Secondary Dial Tone

Secondary dial tone is a system feature to generate a secondary dial tone after the user has
begun dialing an external number. This dial tone is then played until the number dialing and an
external trunk seized.
• Pre-Release 4.0: Secondary dial tone is triggered through the use of the secondary dial tone
short code feature.
• Release 4.0+: The use of this short code feature has been replaced by the Secondary Dial
Tone check box option on ARS forms.
Details
• Telephone Number: Digit which triggers secondary dial tone.
• Default Short Code: 9 (U-Law only)
• Programmable Button Control:
• Release: 1.0+.
Example
For pre-4.0 systems secondary dial tone works in two parts. The following system short code will
trigger secondary dial tone. To use it to trigger secondary dial tone and then continue dialing, other
user, user rights and system short codes should begin with [9].
• Short Code: 9
• Telephone Number: .
• Feature: Secondary Dial Tone
Related links
Short Code Features on page 953

Set Absent Text

This feature can be used to select the user's current absence text. This text is then displayed
to internal callers who have suitable display phones or applications. It doesn't changes the users

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 Set Absent Text

status. The absence text message is limited to 128 characters. Note however that the amount
displayed will depend on the caller's device or application.
The text is displayed to callers even if the user has forwarded their calls or is using follow me.
Absence text is supported across a multi-site network.
Details
• Telephone Number: The telephone number should take the format "y,n,text" where:
- y = 0 or 1 to turn this feature on or off.
- n = the number of the absent statement to use, see the list below:
0 = None. 4 = Meeting until. 8 = With cust. til.
1 = On vacation until. 5 = Please call. 9 = Back soon.
2 = Will be back. 6 = Don't disturb until. 10 = Back tomorrow.
3 = At lunch until. 7 = With visitors until. 11 = Custom.

- text = any text to follow the absent statement.

• Default Short Code:
• Programmable Button Control: Absnt
• Release: 1.0+.
Example
The following short code can be used to turn an absent text message on:
• Short Code: *88
• Telephone Number: "1,5,me on 208"
• Line Group ID: 0
• Feature: SetAbsentText
Example
The following short code could be used to turn this facility off. In the Telephone Number the first 0
is used to turn this facility off and the second 0 is used to select the absent statement "None".
• Short Code: *89
• Telephone Number: "0,0"
• Line Group ID: 0
• Feature: SetAbsentText
Related links
Short Code Features on page 953

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Short Code Features

Set Account Code

This short code feature is used to allow system users to enter a valid account code prior to making
a phone call. Once this short code is set up, any existing account code in the system configuration
can be used in conjunction with it.
This short code feature is essential for allowing analog phone users to enter account codes as
they cannot enter account code through the phone during a call or after dial a number.
Details
• Telephone Number: A valid account code.
• Default Short Code:
• Programmable Button Control: Acct.
• Release: 2.1+.
Example
In this example, N represents any valid account code. For the purpose of this example, we will
imagine the account code to be 1234. Once this short code is created, a user can dial 11*1234#
to get a dial tone for dialing the restricted telephone number or the phone number needing to be
tracked for billing purposes.
• Short code: 11*N#
• Telephone Number: N
• Feature: SetAccountCode
Related links
Short Code Features on page 953

Set Authorization Code

This short code feature is only available on systems configured to use authorization codes. See
Authorization Codes. The feature is used to allow a user to enter a valid authorization code prior to
making a phone call.
This short code feature is essential for allowing analog phone users to enter authorization codes.
Note that the authorization code must be associated with the user or the user rights to which the
user belongs.
Details
• Telephone Number: A valid authorization code.
• Default Short Code:
• Programmable Button Control:
• Release: 3.2+.

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 Set Fallback Twinning Off

Related links
Short Code Features on page 953

Set Fallback Twinning Off

This feature can be used by users to disable fallback twinning operation. This feature requires the
user to have a mobile twinning number set.
Fallback twinning redirects calls to the user’s configured mobile twinning number when the system
cannot detect a connection to the user’s normal registered extension. This feature can be used
without mobile twinning itself being enabled.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
Related links
Short Code Features on page 953

Set Fallback Twinning On

This feature can be used by users to enable fallback twinning operation. This feature requires the
user to have a mobile twinning number set.
Fallback twinning redirects calls to the user’s configured mobile twinning number when the system
cannot detect a connection to the user’s normal registered extension. This feature can be used
without mobile twinning itself being enabled.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
Related links
Short Code Features on page 953

Set Hunt Group Night Service

This feature puts the specified hunt group into Night Service mode.

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Short Code Features

Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Details
• Telephone Number: Hunt group extension number. If left blank, the short code will affect
all hunt groups of which the user is a member.
- The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively. The
service is indicated by setting the service name as the telephone number or action data.
Do not use quotation marks.
• Default Short Code: *20*N#
• Programmable Button Control: HGNS+
• See also: Set Hunt Group Out Of Service, Clear Hunt Group Night Service, Clear Hunt
Group Out Of Service.
• Release: 1.0+.
Example
This short code is a default within the system configuration. N represents the telephone number of
the hunt group to be placed into "Night Service" mode. For example, when *20*201# is dialed, the
hunt group associated with extension 201 will be placed into "Night Service" mode.
• Short Code: *20*N#
• Telephone Number: N
• Feature: SetHuntGroupNightService
Related links
Short Code Features on page 953

Set Hunt Group Out Of Service

This feature manually puts the specified hunt group into Out of Service mode. If a time profile has
also been defined to control hunt group night service, the action may vary:
• Set Hunt Group Out of Service can be used to override a time profile and change a hunt
group from night service to out of service.
Details
• Telephone Number: Hunt group extension number. For Release 4.0+, if left blank, the
short code will affect all hunt groups of which the user is a member.
• Default Short Code:

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 Set Inside Call Seq

• Programmable Button Control: HGOS+

• Release: 1.0+.
Example
Below is a sample short code using the Set Hunt Group Out Of Service feature. N represents
the telephone number of the hunt group to be placed into "Out of Service" mode. For example,
when *56*201# is dialed, the hunt group associated with extension 201 will be placed into "Out of
Service" mode.
• Short Code: *56*N#
• Telephone Number: N
• Feature: SetHuntGroupOutOfService
Related links
Short Code Features on page 953

Set Inside Call Seq

This feature allows the user to select the ringing used on their analog extension for internal calls.
Details
• Telephone Number: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.
• Default Short Code:
• Programmable Button Control: ICSeq
• See also: Set Ringback Seq, Set Inside Call Seq.
• Release: 1.0+.
Example
This Short Code allows a user to change their inside call pattern. N represents the number
corresponding to the Call Sequence the user wishes to choose.
• Short Code: *80*N#
• Telephone Number: N
• Feature: SetInsideCallSeq
Related links
Short Code Features on page 953

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Short Code Features

Set Mobile Twinning Number

This short code feature can be used to set a mobile twinning number. The destination can be any
external number the user is able to dial normally. It should include any prefix if necessary.
Details
• Telephone Number: Twinning destination.
• Default Short Code:
• Programmable Button Control:
• See also: Set Mobile Twinning On, Set Mobile Twinning Off, Mobile Twinned Call Pickup.
• Release: 3.2+.
Related links
Short Code Features on page 953

Set Mobile Twinning On

This short code feature turns on the user's mobile twinning. It requires a mobile twinning number
to have been set for the user. That can be done through using the Set Mobile Twinning Number
short code feature or through the User | Twinning tab within Manager.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• See also: Set Mobile Twinning Off, Set Mobile Twinning Number, Mobile Twinned Call
Pickup.
• Release: 3.2+.
Related links
Short Code Features on page 953

Set Mobile Twinning Off

This short code feature turns off the user's mobile twinning.
Details
• Telephone Number:
• Default Short Code:

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 Set No Answer Time

• Programmable Button Control:

• See also: Set Mobile Twinning On, Set Mobile Twinning Number, Mobile Twinned Call
Pickup.
• Release: 3.2+.
Related links
Short Code Features on page 953

Set No Answer Time

This short code feature allows the user to change their No Answer Time (User | Telephony | Call
Settings).
Details
• Telephone Number: Time in seconds.
• Default Short Code:
• Programmable Button Control: NATim
• See also: Set Wrap Up Time.
• Release: 1.0+.
Example
This short code allows a user to change the length of time they have to answer the phone before
it goes to divert or voicemail. N represents the number of seconds. For example, if a user wants to
set the no answer time to 15 seconds, the following information needs to be entered: *81*15#.
• Short Code: *81*N#
• Telephone Number: N
• Feature: SetNoAnswerTime
Related links
Short Code Features on page 953

Set Outside Call Seq

This feature allows the user to select the ringing used on their analog extension for external calls.
Details
• Telephone Number: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.

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Short Code Features

• Default Short Code:

• Programmable Button Control: OCSeq
• See also: Set Ringback Seq, Set Outside Call Seq.
• Release: 1.0+.
Example
This short code allows a user to change the ringing tone for an external call. N represents the
number corresponding to the Call Sequence the user wishes to choose.
• Short Code: *81*N#
• Telephone Number: N
• Feature: SetOutsideCallSeq
Related links
Short Code Features on page 953

Set Ringback Seq

This feature allows the user to select the ringing used on their analog extension for ringback calls.
Details
• Telephone Number: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.
• Default Short Code:
• Programmable Button Control: RBSeq
• See also: Set Outside Call Seq, Set Inside Call Seq.
Example
This short code allows a user to change the ringing tone for a ringback call. N represents the
number corresponding to the ring tone the user wishes to choose.
• Short Code: *81*N#
• Telephone Number: N
• Feature: SetRingbackSeq
Related links
Short Code Features on page 953

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 Set Time Profile

Set Time Profile

You can manually override a time profile. The override settings allow you to mix timed and manual
settings.
Five short codes can be configured.
Short Code Name Description
Set Time Profile Timed Operation No override. The time profile operates as
configured.
Set Time Profile Active Until Next Timed Inactive Use for time profiles with multiple intervals. Select
to make the current timed interval active until the
next inactive interval.
Set Time Profile Inactive Until Next Timed Active Use for time profiles with multiple intervals. Select
to make the current active timed interval inactive
until the next active interval.
Set Time Profile Latch Active Set the time profile to active. Timed inactive periods
are overridden and remain active.
Set Time Profile Latch Inactive Set the time profile to inactive. Timed active periods
are overridden and remain active.

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

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Short Code Features

Details
• Telephone Number: Time profile name.
•
• Default Short Code: No.
• Programmable Button Control: Yes: Time Profile
Related links
Short Code Features on page 953

Set Wrap Up Time

Allows users to change their Wrap-up Time (User | Telephony | Call Settings) setting.
• Other phones or applications monitoring the user's status will indicate the user as still being
busy (on a call).
• Hunt group calls will not be presented to the user.
• If the user is using a single line set, direct calls also receive busy treatment. If the user is
using a mutli-line set (multiple call appearances), direct calls to them will ring as normal.
• It is recommended that this option is not set to less than the default of 2 seconds. 0 is used to
allow immediate ringing.
• For users set as an CCR Agent, the After Call Work Time (User | Telephony | Supervisor
Settings) setting should be used.
Details
• Telephone Number: Time in seconds.
• Default Short Code:
• Programmable Button Control: WUTim
• See also: Set No Answer Time.
• Release: 1.0+.
Example
N represents the number of seconds. For example, if a user wants to set her/his wrap up time to
8 seconds, this user would dial *82*5#. This short code is useful in a "call center" environment
where users may need time to log call details before taking the next call. If set to 0 the user does
not receive any calls. It is recommended that this option is not set to less than the default of 2
seconds.
• Short Code: *82*N#
• Telephone Number: N
• Feature: SetWrapUpTime

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 Speed Dial

Related links
Short Code Features on page 953

Speed Dial
Each system directory and personal directory number stored in the configuration can be optionally
assigned an index number. That index number can then be used by M-Series and T-Series phone
users to dial the directory number. This short code feature allows the creation of short codes
to perform the same function. However, the short code is diallable from any type of telephone
extension on the system.
For example:
• If Feature 0 is followed by a 3-digit index number in the range 000 to 999, the system
directory record with the matching index number is dialed.
• If Feature 0 is followed by * and a 2-digit index number in the range 00 to 99, the personal
directory record with the matching index number is dialed. Alternatively Feature 0 can be
followed by 00# to 99#. Note: Release 10.0 allows users to have up to 250 personal directory
entries. However, only 100 of those can be assigned index numbers.
Details
• Telephone Number: System directory entry index number (000 to 999) or personal
directory entry index number (00 to 99).
• Default Short Code:
• Programmable Button Control:
• Release: 8.1.
Example
Using the example below, a user is able to dial *0 and then either a 2 digit code for an indexed
personal directory entry or a 3 digit code for an indexed system directory entry.
• Short Code: *0N#
• Telephone Number: N
• Feature: Speed Dial
Related links
Short Code Features on page 953

Shutdown Embedded Voicemail

Allows the Embedded Voicemail service provided by an Avaya memory card in a control unit to be
shut down. To restart the service, a Startup Embedded Voicemail short code should be used.

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Short Code Features

The short code has the following effects:

1. Immediately disconnect all current users within Embedded Voicemail. This is not a polite
shutdown.
2. Mark the Embedded Voicemail as inactive so that it will not receive any new calls.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• Release: 4.0+ (Added in the Release 4.0 Q2 2007 maintenance release).
Related links
Short Code Features on page 953

Stamp Log
The stamp log function is used to insert a line into any System Monitor trace that is running. The
line in the trace indicates the date, time, user name and extension plus additional information. The
line is prefixed with LSTMP: Log Stamped and a log stamp number. When invoked from a Avaya
phone with a display, Log Stamped# is also briefly displayed on the phone. This allows users to
indicate when they have experienced a particular problem that the system maintainer want them
to report and allows the maintainer to more easily locate the relevant section in the monitor trace.
The log stamp number is set to 000 when the system is restarted. The number is then
incremented after each time the function is used in a cycle between 000 and 999. Alternately
if required, a specific stamp number can be assigned to the button or short code being used for
the feature.
Details
• Telephone Number: Optional. If not set, a number in the sequence 000 to 999 is
automatically used. If set, the number set is used.
• Default Short Code: *55
• Programmable Button Control: Stamp Log
• Release: 8.1+
Related links
Short Code Features on page 953

Startup Embedded Voicemail

Restarts the Embedded Voicemail service provided by an Avaya Memory in a control unit.

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 Suspend Call

Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
• Release: 6.0+
Related links
Short Code Features on page 953

Suspend Call
This feature uses the Q.931 Suspend facility. It suspends the incoming call at the ISDN exchange,
freeing up the ISDN B channel. The call is placed in exchange slot 0 if a slot number is not
specified.
Details
• Telephone Number: Exchange slot number or blank (slot 0).
• Default Short Code:
• Programmable Button Control: Suspe
• See also: Resume Call.
• Release: 1.0+.
Related links
Short Code Features on page 953

Suspend CW
This feature uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange
and answer the call waiting. The call is placed in exchange slot 0 if a slot number is not specified.
Only available when supported by the ISDN exchange.
Details
• Telephone Number: Exchange slot number or blank (slot 0).
• Default Short Code: *28*N# (A-Law only) (not on Server Edition)
• Programmable Button Control: SusCW
• See also: Resume Call.
• Release: 1.0+.

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Short Code Features

Example
Sample short code using the Suspend CW feature.
• Short Code: *28*N#
• Feature: Suspend CW
Related links
Short Code Features on page 953

Start After Call Work

This feature can be used by users who have been configured as CCR agents. It allows them
to dial a short code to enter the After Call Work (ACW) state as reported by the Customer Call
Reporter (CCR) application.
• CCR is not supported in IP Office release 9.1 and later.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control: ACWrk
• See also: Clear After Call Work.
• Release: 4.2 4Q 2008 Maintenance release+.
Related links
Short Code Features on page 953

Toggle Calls
This feature cycles through each call that the user has on hold on the system. This feature is
useful when a user with a single-line telephone has several calls on hold and needs to respond to
each one in turn.
Details
• Telephone Number:
• Default Short Code: *29
• Programmable Button Control: Toggl
• Release: 1.0+.

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 Unpark Call

Example
Below is sample short code using the Toggle Calls feature.
• Short Code: *29
• Feature: ToggleCalls
Related links
Short Code Features on page 953

Unpark Call
Retrieve a parked call from a specified system park slot.
Details
• Telephone Number: System park slot number.
• Default Short Code: *38*N#
• Programmable Button Control: Ride
• See also: Call Park.
• Release: 1.0+.
Example
Below is a sample short code using the Unpark Call feature. N represents the park slot number in
which the call you want to retrieve was parked. For example, if a user parked a call to slot number
9, you can retrieve that call by dialing *38*9#.
• Short Code: *38*N#
• Telephone Number: N
• Feature: Unpark Call
Related links
Short Code Features on page 953

Voicemail Collect
This feature connects to the voicemail system. The telephone number field is used to indicate the
name of the mailbox to be accessed, for example "?Extn201" or "#Extn201".
• ? indicates 'collect messages'.
• # indicates 'leave a message'. It also instructs the voicemail server to give a brief period
of ringing before connecting the caller. This is useful if the short code is used for functions
such as call transfers as otherwise the voicemail server can start playing prompts before the
transfer is completed. However, the # can be omitted for immediate connection if required.

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Short Code Features

• " " quotation marks must be used to enclose any information that needs to be sent to the
voicemail server as is. Any text not enclosed by quote marks is checked by the telephone
system for short code character matches which will be replaced before being sent to the
voicemail server.
- Manager automatically adds quotation marks to the Telephone Number field if they are
not added manually. Care should be taken to ensure that special characters that you want
replaced by the telephone system, such as U, N or X, are not enclosed by the quotation
marks. For scenarios where the telephone number only contains short code characters,
add an empty pair of quotation marks, for example ""N.
When using Voicemail Pro, names of specific call flow start points can directly access those start
points via a short code. In these cases, ? is not used and # is only needed if ringing is required
before the start point's call flow begins.
Short codes using the Voicemail Collect feature, with either "Short Codes.name" and
"#Short Codes.name" records in the Telephone Number field are automatically converted
to the Voicemail Node feature and name.
CallPilot voicemail is used for IP Office Branch deployments with CS 1000. Users can access their
CallPilot voicemail by dialing the Voicemail Collect short code. For access to CallPilot voicemail
from an Auto Attendant, set a Normal Transfer action to point to the CallPilot number.
Details
• Telephone Number: See the notes above.
• Default Short Code: *17
• Programmable Button Control: VMCol
• See also: Voicemail On, Voicemail Off, Voicemail Node.
• Release: 1.0+.
Example: Retrieve Messages from Specific Mailbox
This short code allows a user to retrieve messages from the mailbox of the hunt group 'Sales'.
This usage is not supported on Voicemail Pro running in Intuity emulation mode unless a custom
call flow has been created for the hunt group, refer to the Voicemail Pro help.
• Short Code: *89
• Telephone Number: "?Sales"
• Feature: VoicemailCollect
Example: Record Message to Specific Mailbox
To allow users to deposit a message directly to Extn201's Voicemail box. This short code is useful
when you know the person is not at her/his desk and you want to immediately leave a message
rather than call the person and wait to be redirected to voicemail.
• Short Code: *201
• Telephone Number: "#Extn201"
• Feature: VoicemailCollect

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 Voicemail Node

Example: Accessing a Specific Voicemail Pro Module

This short code can be used in instances where you have a conference bridge set up on the
system and a module has been created via Voicemail Pro to access this conference bridge. A
short code can be created for internal access to the module. In the sample short code below, the
telephone number field contains the name of the module. In this example, if a short burst of ringing
is required before connecting the module, "#conferenc" would be used as the telephone number.
• Short Code: *100
• Telephone Number: "conferenc"
• Feature: VoicemailCollect
Related links
Short Code Features on page 953

Voicemail Node
Similar to Voicemail Collect but used for calls being directed to a Voicemail Pro Short Codes start
point. Useful if you have set up a short code start point with Voicemail Pro and want to give direct
internal access to it.
Details
• Telephone Number: Voicemail Pro Short Code start point name without quotation marks.
• Default Short Code:
• Programmable Button Control:
• See also: Voicemail Collect.
• Release: 2.0+.
Example
Having created a short codes start point call flow called Sales, the following system short code
can be used to route calls to that call flow:
• Short Code: *96
• Telephone Number: Sales
• Feature: VoicemailNode
Related links
Short Code Features on page 953

Voicemail On
This feature enables the user's voicemail mailbox to answer calls which ring unanswered or arrive
when the user is busy.

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Short Code Features

Details
• Telephone Number: None.
• Default Short Code: *18
• Programmable Button Control: VMOn
• See also: Voicemail Off.
• Release: 1.0+.
Example
This short code can be used to toggle the feature on.
• Short Code: *18
• Feature: VoicemailOn
Related links
Short Code Features on page 953

Voicemail Off
This feature disables the user's voicemail mailbox box from being used to answer calls. It does not
disable the voicemail mailbox being used as the target for other functions such as call recording or
messages forwarded from other mailboxes.
Details
• Telephone Number: None.
• Default Short Code: *19
• Programmable Button Control: VMOff
• See also: Voicemail On.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *19
• Feature: VoicemailOff
Related links
Short Code Features on page 953

Voicemail Ringback On
This feature enables voicemail ringback to the user's extension. Voicemail ringback is used to
call the user when they have new voicemail messages. The ringback takes place each time the

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 Voicemail Ringback Off

extension is used. This feature is useful for users who do not have voicemail light/button indicators
on their telephone.
If the user has been configured to receive message waiting indication for any hunt groups, a
separate voicemail ringback will occur for each such group and for the users own mailbox.
Details
• Telephone Number:
• Default Short Code: *48
• Programmable Button Control: VMRB+
• See also: Voicemail Ringback Off.
• Release: 1.0+. For Release 3.2, the Voicemail On and Voicemail Ringback On short code
features toggled. For Release 4.0 and higher, they no longer toggle.
Example
This short code can be used to turn the feature on.
• Short Code: *48
• Feature: VoicemailRingbackOn
Related links
Short Code Features on page 953

Voicemail Ringback Off

This feature disables voicemail ringback to the user's extension.
Details
• Telephone Number:
• Default Short Code: *49
• Programmable Button Control: VMRB-
• See also: Voicemail Ringback On.
• Release: 1.0+.
Example
Below is a sample of the short code setup.
• Short Code: *49
• Feature: VoicemailRingbackOff
Related links
Short Code Features on page 953

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Short Code Features

Whisper Page
This feature allows you to intrude on another user and be heard by them without being able to
hear the user's existing call which is not interrupted.
For example: User A is on a call with user B. When user C intrudes on user A, they can be heard
by user A but not by user B who can still hear user A. Whisper page can be used to talk to a user
who has enabled private call.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Telephone Number: Target extension number.
• Default Short Code:
• Programmable Button Control: Whisp.
• See also: Call Intrude, Call Listen, Coaching Intrusion, Dial Inclusion.
• Release: 8.0+.
Related links
Short Code Features on page 953

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Part 12: Button Programming

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Chapter 102: Button Programming
Overview

This section provides an overview of system actions that can be assigned to programmable buttons
on Avaya phones.
Button assignment can be done through the system configuration using IP Office Manager and IP
Office Web Manager. If only button programming changes are required, the configuration changes
can be merged back to the system without requiring a reboot.
Users can also do their own button programming using the user portal application or, on some
phones, through the phone's menu. However, users can only program a limited set of functions and
cannot override appearance buttons and buttons set through user rights templates.
• Appearance Functions
The functions Call Appearance, Bridged Appearance, Coverage and Line Appearance are
collectively known as "appearance functions". For full details of their operation and usage, see
Appearance Buttons on page 1159.
• Phone Support
Note that not all functions are supported on all phones with programmable buttons. Where
possible exceptions, have been indicated. Those buttons will typically play an error tone when
used on that phone. Programming of those features however is not restricted as users may hot
desk between different types of phones, including some where the feature is supported.
• Status Indication
Actions that use status feedback are only supported on buttons that provide that feedback
through lamps or icons.
Related links
Programming Buttons with IP Office Manager on page 1043
Interactive Button Menus on page 1044
Label Templates on page 1044

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 Programming Buttons with IP Office Manager

Programming Buttons with IP Office Manager

This process edits the programmable buttons for individual users.
• You can also use user rights to create a set of programmable buttons that are simultaneously
applied to multiple users. See Configuring User Rights on page 740.
Procedure

1. Select the User required to display their configuration details.

2. Select Button Programming. The number of buttons displayed is based on the phone
associated with the user when the configuration was loaded. This can be overridden by
selecting Display All Buttons. This may be necessary for users who switch between
different phones using hot desking or have an expansion unit attached to their phone.

3. For the required button, either select the button and then click Editor double-click the
button.
4. Edit the settings as required. Use the .... button to display the menu for selecting the
required button action. Select the action and set the action data, then click OK.

5. Click OK. Repeat for any other buttons.

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Button Programming Overview

6. Click OK.
Related links
Button Programming Overview on page 1042

Interactive Button Menus

For certain functions, on display phones where a button has been configured without a specific
number, a menu for number entry is displayed. The menu includes a Dir option for selecting a
number from the directories held by the system.
Functions that use the interactive menu are:
Feature Directory lists... Feature Directory lists...
Automatic Intercom Users Follow Me Here Cancel Users
Acquire Call/Call Steal Users Follow Me Here Users
Call Forwarding All Users Follow Me To Users
Call Intrude Users Forward Number Users/Groups
Call Park To Other Users Forward Busy Number Users/Groups
Extension
Dial Inclusion Users Group Paging Users/Groups
Dial Intercom Users Leave Word Calling Users/Groups
Directed Call Pickup Users/Groups Priority Calling Users/Groups

User and Group buttons can be used to indicate the required user or hunt group only if those
buttons are on an associated button module. User and Group buttons on the users extension are
not accessible while the interactive button menu is being displayed.
For functions supported across a multi-site network, the directory will include remote users and
advertised hunt groups.
For M-Series and T-Series phone, the volume buttons are used to scroll through the list of
matching names. If this is done during a call or while a call is alerting, this will also adjust the call
or ring volume.
Related links
Button Programming Overview on page 1042

Label Templates
A zip file is available containing Word document templates for the paper programmable key
labels used on various phones supported by the system. Two templates are provided, one for A4

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 Label Templates

size paper, the other for US Letter sized paper. See https://ipofficekb.avaya.com/businesspartner/
ipoffice/user/dsstemplate/index.htm.
For 1400 and 1600 phones, a number of tools and perforated printable labels are available. For
further details visit http://support.avaya.com and search for information on DESI. Alternatively, visit
http://www.desi.com.
Related links
Button Programming Overview on page 1042

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Chapter 103: Button Programming Actions

The following sections provide details for each of the button actions supported by system. Note that
this does not include buttons on phones on a system running in Partner Edition mode.
For each, the following details are listed:
• Action - Indicates the selection path to the action from within the list of actions displayed in
Manager.
• Action Data - Indicates the type of data required by the action. For some actions no data
is required while for others action data may be optional. The option to enter the data after
pressing the button is not available for all phones, see Interactive Button Menus.
• Default Label - This is the default text label displayed on phones which provide a display area
next to programmable buttons. Alternate labels can be specified in the system configuration or
entered by the phone user (refer to the telephone user guide). Note that for buttons with action
data set, the action data may also be displayed as part of the default label. Depending on the
display capacity of the particular phone, either a short or long label is displayed.
• Toggles - Indicates whether the action toggles between two states, typically on or off.
• Status Indication - Indicates whether the button provides status indication relevant to the
feature if the button has status lamps or display. If the Status Indication is listed as Required
it indicates that the button action is only supported on programmable buttons that can provide
status indication.
• User Admin - This item indicates that users with a Self-Administer button can assign the action
to other buttons themselves.
• Phone Support - This is only a general indication of support or otherwise of an action by
phones within particular series. On phones with 3 or less programmable buttons those button
can only be used for the Call Appearance action. In addition some actions are only supported
on phones where the programmable buttons provide status indication and or a display for data
entry once the feature is invoked.

Button Programming Actions Summary

The following tables list the actions available for programmable buttons on system.

• Login Code Required Some function may require the user to enter their log in code. This
typically applies when the action data is left blank for entry when the button is pressed.

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 Button Programming Actions Summary

General
Action Action Data Default Label
Dial Any number. Dial
Group "Group name" in quote marks. <Group name>
User "User name" in quote marks. <User name>

Appearance
Action Action Data Default Label
Appearance None. a=
Bridged Appearance User name and call appearance <user name><appearance label>
button number.
Coverage Appearance User name. <user name>
Line Appearance Line appearance ID. Line

Emulation
Action Action Data Short Label Long Label
Abbreviated Dial Any number. AD Abbreviate Dial
Abbreviated Dial Pause None. Pause –
Abbreviated Dial None. Prog –
Program
Abbreviated Dial Stop None. Stop –
Absent Message None. None. None.
Account Code Entry Account code or blank Acct Account Code
for entry when pressed.
ACD Agent Statistics None. Stats –
ACD Stroke Count None. Count –
AD Special Function None. Mark –
Mark
AD Special Function None. Wait –
Wait
AD Special Functions None. Sfunc –
AD Suppress None. Spres Suppress Digits
Automatic Callback None. AutCB Auto Callback
Automatic Intercom User number or name. Iauto Auto Intercom
Call Forwarding All Any number or blank for CFrwd Call Forward All
entry when pressed.
Call Park Park slot ID CPark Call Park
(alphanumeric) or blank
for menu of slots in use.
Table continues…

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Button Programming Actions

Action Action Data Short Label Long Label

Call Park To Other User number. RPark Call Park to Other
Extension
Call Pickup None. CpkUp Call Pickup Any
Cancel Leave Word None. CnLWC –
Calling
Consult None. Cnslt –
Dial Intercom User number or name Idial Auto Intercom
or blank for entry when
pressed.
Directed Call Pickup User number or name or DpkUp Call Pickup
group number or name
or or blank for entry
when pressed..
Directory None. Dir –
Drop None. Drop Drop Call
Emergency View None. 911–View or EView
Group Paging User or group number or GrpPg Page
name or blank for entry
when pressed.
Headset Toggle None or FF HdSet –
Inspect None. Inspt –
Internal Auto-Answer None. HFAns Auto Answer
Leave Word Calling None. LWC –
Manual Exclude None. Excl –
Priority Calling None. Pcall –
Ringer Off None. RngOf Ringer Off
Self-Administer Blank or 1 or 2 Admin Self Administer
Send All Calls None. SAC Send All Calls
Stored Number View None. BtnVu –
Time of Day None. TmDay –
Timer None. Timer –
Twinning None. Twinning Twinning
Visual Voice None. Voice Voice

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 Button Programming Actions Summary

Advanced
Action Action Data Category Short Label Long Label
Acquire Call User number or Call Acquir Acquire
blank for last call
transferred.
Break Out System name or IP Dial BkOut Breakout
address or blank
for selection when
pressed.
Busy None. Busy Busy –
Busy On Held 0 (off) or 1 (on). Busy BusyH –
Call Intrude User number or Call Intru Call Intrude
blank for entry
when pressed.
Call List None. Call LIST –
Call Listen User number. Call Listn Listen
Call Log None. Call Call Log
Call Pickup Any None. Call PickA Pickup Any
Call Pickup Group Group number or Call PickG Pickup Group
name.
Call Pickup Group number or Call PickM Pickup Members
Members name.
Call Queue User number. Call Queue Queue
Call Record None. Call Recor Record
Call Screening None. Call CallScreen Call Screening
Call Steal User number or Call Steal –
blank for last call
transferred.
Call Waiting Off None. Call CWOff –
Call Waiting On None. Call CWOn –
Call Waiting None. Call CWSus –
Suspend
Cancel All None. Call FwdOf Call Forward Off
Forwarding
Cancel Ring Back None. Miscellaneous RBak- –
When Free
Channel Monitor Channel number. Call ChMon –
Clear Call None. Call Clear Clear
Clear CW None. Call ClrCW –
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Clear Hunt Group Group number. Call HGNS- –
Night Service
Clear Hunt Group Group number. Call HNOS- –
Out Of Service
Clear Quota "Service name" Call Quota –
within quote marks
or "" for all
services.
Coaching Intrusion User number or Call Coach Coaching
name or blank Intrusion
for entry when
pressed.
Conference Invoke the Call Conf –
conference
process. (M and
T-Series phones
only)
Conference Add None. Call Conf+ Conference Add
Conference Meet Conference name Call CnfMM Conf. Meet Me
Me or number.
Dial 3K1 Any number. Dial D3K1 Dial 3K1
Dial 56K Any number. Dial D56K Dial 56K
Dial 64K Any number. Dial D64K Dial 64K
Dial CW User number. Dial DCW Dial Call Waiting
Dial Direct User number or Dial Dirct Auto Intercom
name or blank
for entry when
pressed.
Dial Emergency Any number. Dial Emrgy Dial Emergency
Dial Inclusion User number or Dial Inclu Dial Inclusion
name or blank
for entry when
pressed.
Dial Paging User or group Dial Page Page
number or name
or blank for entry
when pressed.
Dial Physical Extn Extension port Dial PhyEx Dial Physical Extn
by Number Base Extension
number.
Table continues…

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 Button Programming Actions Summary

Action Action Data Category Short Label Long Label

Dial Physical Extn Extension port ID Dial DialP Dial Extn by Id
by Id number. (Release
1.4+)
Dial Speech Any number. Dial DSpch Dial Speech
Dial V110 Any number. Dial DV110 Dial V110
Dial V120 Any number. Dial DV120 Dial V120
Dial Video Any number. Dial Dvide Dial Video
Display Msg Command string. Dial Displ –
Do Not Disturb None Do Not Disturb NoAI No Auto Int Calls
Auto-Intercom
Deny
Do Not Disturb Any number. Do Not Disturb DNDX+ –
Exception Add
Do Not Disturb Any number. Do Not Disturb DNDX- –
Exception Delete
Do Not Disturb Off None. Do Not Disturb DNDOf –
Do Not Disturb On None. Do Not Disturb DNDOn Do Not Disturb
Extn Login None. Extension Login Login
Extn Logout None. Extension Logof Logout
Flash Hook None. Miscellaneous Flash Flash Hook
Follow Me Here User number. Follow Me Here+ Follow Me Here
Follow Me Here User number or Follow Me Here- Follow Me Here-
Cancel blank for entry
when pressed.
Follow Me To User name or user Follow Me FolTo Follow Me To
number or blank
for entry when
pressed.
Forward Hunt None. Forward FwdH+ –
Group Calls On
Forward Hunt None. Forward FwdH- Fwd HG Calls
Group Calls Off
Forward Number Any number or Forward FwdNo Fwd Number
blank for entry
when pressed.
Forward On Busy Any number or Forward FwBNo Fwd Busy
Number blank for entry Number
when pressed.
Forward On Busy None. Forward FwBOf –
Off
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Forward On Busy None. Forward FwBOn Fwd Busy
On
Forward On No None. Forward FwNOf –
Answer Off
Forward On No None. Forward FwNOn Fwd No Answer
Answer On
Forward None. Forward FwUOf –
Unconditional Off
Forward None. Forward FwUOn Fwd
Unconditional On Unconditional
Group Listen On None. Extension GroupListenOn –
Hold Call ISDN Exchange Hold Hold –
slot number.
Hold CW None. Hold HoldCW –
Hold Music None. Hold Music Hold Music
Hunt Group Group number or Hunt Group HGDis
Disable name or blank for
all groups.
Hunt Group Enable Group number or Hunt Group HGEna HG Enable
name or blank for
all groups.
Last Number Redial the last Call Again –
Redial number dialed.
(M and T-Series
phones only)
MCID Activate None. Miscellaneous MCID Malicious Call
Monitor Analogue Line appearance Voicemail TrkMW Trunk MWI
Trunk MWI ID.
Off Hook Station None. Miscellaneous OHStn –
Pause Recording None. Call PauseRec Pause Recording
Priority Call User number or Call PCall Priority Call
name.
Private Call None. (Release Call PrivC Private Call
4.0+)
Relay Off 1 or 2. Relay Rely- –
Relay On 1 or 2. Relay Rely+ Relay On
Relay Pulse 1 or 2. Relay Relay Relay Pulse
Resume Call ISDN Exchange Call Resum –
slot number.
Table continues…

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 Button Programming Actions Summary

Action Action Data Category Short Label Long Label

Retrieve Call ISDN Exchange Call Retriv –
slot number.
Ring Back When None. Miscellaneous RBak+ Auto Callback
Free
Set Absent Text String for selected Set Absnt Absence Text
message and
custom text.
Set Account Code Blank or valid Set Acct Account Code
account code.
(Release 2.1+)
Set Hunt Group Group number. Set HGNS+ HG Night Service
Night Service
Set Hunt Group Group number. Set HGOS+ HG Out of Service
Out Of Service
Set Inside Call Seq Value 0 to 10. Set ICSeq –
Set Night Service Group number. Set SetNSG HG NS Group
Group (Release 4.2+)
Set No Answer Time in seconds Set NATim No Answer Time
Time (range 6 to 99999).
Set Outside Call Value 0 to 10. Set OCSeq –
Seq
Set Out of Service Group number. Set SetOOSG HG OS Group
Group (Release 4.2+)
Set Ringback Seq Value 0 to 10. Set RBSeq –
Set Wrap Up Time Time in seconds Set WUTim Wrap-up Time
(range 0 to 99999).
Speed Dial Initiate the speed Dial SpdDial –
dial selection
process. (M and
T-Series phones
only)
Stamp Log None. Miscellaneous StmpL Stamp Log
Suspend Call ISDN Exchange Suspend Suspe –
slot number.
Suspend CW ISDN Exchange Suspend SusCW –
slot number.
Toggle Calls None. Call Toggl –
Transfer Initiate the call Call Xfer –
transfer process.
(M and T-Series
phones only)
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Unpark Call Park slot ID Call Ride –
(alphanumeric).
Voicemail Collect See notes. Voicemail VMCol VMail Collect
Voicemail Off None. Voicemail VMOff –
Voicemail On None. Voicemail VMOn VMail On
Voicemail Ringback None. Voicemail VMRB- –
Off
Voicemail Ringback None. Voicemail VMRB+ VMail Ringback
On
Whisper Page User number or Call Whisp Whisper Page
name or blank
for entry when
pressed.
Channel Monitor Channel Call ChMon -

911-View
See Emergency View on page 1103.

Abbreviated Dial
This function allows quick dialing of a stored number.
Details
• Action: Emulation | Abbreviated Dial.
• Action Data:
- Full Number The number is dialed.
- Partial Number The partial number is dialed and the user can then complete dialing the
full number.
• Default Label: AD or Abbreviate Dial.
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.

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 Abbreviated Dial Pause

- 1400 Series and 1600 Series.

- M-Series and T-Series.

Abbreviated Dial Pause

Supported for CTI emulation only.
Allows a user to enter a pause character when programming an abbreviated dial.
Details
• Action: Emulation | Abbreviated Dial Pause.
• Action Data: None.
• Default Label: Pause.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Abbreviated Dial Program

Supported for CTI emulation only.
Allows a user to program abbreviated dialing numbers against other programmable buttons. This
function cannot be used to overwrite call appearance buttons.
Details
• Action: Emulation | Abbreviated Dial Program.
• Action Data: None.
• Default Label: Prog.
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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Button Programming Actions

Abbreviated Dial Stop

Supported for CTI emulation only.
Allows a user to enter a stop character when programming an abbreviated dial.
Details
• Action: Emulation | Abbreviated Dial Stop.
• Action Data: None.
• Default Label: Stop.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Absent Message
This feature allows to select the user's current absence text. See Set Absent Text on page 1135.

Account Code Entry

Enter an account code for a call. This button can be used before dialing a number or during a call.
Details
• Action: Emulation | Account Code Entry.
• Action Data: Optional. If an code is set it must match an account code set in the account
codes list. If no account code is set, the phone display will request entry of a valid code. This
option is not supported on XX02 phones and the T7000 phone.
• Default Label: Acct or Account Code.
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.

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 ACD Agent Statistics

- 1400 Series and 1600 Series.

- M-Series and T-Series.
- 1100 Series and 1200 Series.

ACD Agent Statistics

Supported for CTI emulation only.
Details
• Action: Emulation | ACD Agent Statistics.
• Action Data: None.
• Default Label: Stats.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

ACD Stroke Count

Supported for CTI emulation only.
Details
• Action: Emulation | ACD Stroke Count.
• Action Data: None.
• Default Label: Count.
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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Button Programming Actions

Acquire Call
See Call Steal on page 1078.

AD Special Functions
Supported for CTI emulation only.
Allows a user to enter a special character (mark, pause suppress, wait) when entering an
abbreviated dial.
Details
• Action: Emulation | AD Special Functions.
• Action Data: None.
• Default Label: Sfunc.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

AD Special Function Mark

Supported for CTI emulation only.
Allows a user to enter a mark character when programming abbreviated dial.
Details
• Action: Emulation | AD Special Function Mark.
• Action Data: None.
• Default Label: Mark.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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 AD Special Function Wait

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

AD Special Function Wait

Supported for CTI emulation only.
Allows a user to enter a Wait for Dial Tone character when programming an abbreviated dial.
Details
• Action: Emulation | AD Special Function Wait.
• Action Data: None.
• Default Label: Wait.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

AD Suppress
Suppresses the display of dialed digits on the telephone display. Dialed digits are replaced with an
s character.
Details
• Action: Emulation | AD Suppress.
• Action Data: None.
• Default Label: Spres or Suppress Digits.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

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Button Programming Actions

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

After Call Work

This button is used by users configured as a Customer Call Reporter (CCR) Agent (User |
Telephony | Supervisor Settings) and working with the CCR application. It shows the CCR
agent their current After Call Work (ACW) status and allow them to manually change status. While
in ACW state, the agent will not receive hunt group calls.
CCR Agents can be automatically put into and taken out of ACW by the system if the user is
configured for Automatic After Call Work (User | Telephony | Supervisor Settings). Those users
must have an After Call Work button.
Note:
CCR is not supported in IP Office release 9.1 and later.
Details
• Action: Advanced | Miscellaneous | After Call Work
• Action Data: None.
• Default Label: ACWrk or After Call Work.
• Toggles: Yes.
• Status Indication: Yes. Required.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

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 Appearance

Appearance
Creates a call appearance button. This can be used to answer and make calls. Users with multiple
call appearance buttons can handle multiple calls. For details, see Call Appearance Buttons on
page 1161.
Call appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Appearance buttons can be set with a ring delay if required or to not ring. This does not affect
the visual alerting displayed next to the button. The delay uses the user's Ring Delay (User >
Telephony > Multi-line Options) setting.
Details
• Action: Appearance | Appearance.
• Action Data: Optional text label.
• Default Label: a=.
• Toggles: No.
• Status Indication: Yes, required.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
Virtual Call Appearances
T7000, T7100, M7100 and M7100N phones support virtual call appearance button operation.
Virtual call appearance operation is similar to an analog phone with call waiting enabled. However,
it does not use the call waiting on/off settings, instead it uses call appearance buttons.
The number of virtual call appearances is set by the call appearance buttons programmed in
the user's settings. These must be programmed as a single block start from button 1. It is
recommended that only a maximum of 3 call appearances are used, however the user must have
at least 1 call appearance programmed in order to make and receive calls.
Virtual Call Appearance Usability
If the user goes off-hook, they are connected to the alerting call if any, else to dial tone in order to
make an outgoing call. This uses one of their virtual call appearance buttons.
With a call connected:
• If another call arrives on another virtual call appearance, the user will hear a call waiting tone
on the set. The display, if the phone has one, will switch between details of the current and
the waiting caller.
• If the user presses Hold, the connected call is placed on hold and:

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Button Programming Actions

If there are any available virtual call appearances, dial tone is heard. This allows the user to make
a call or to use short codes that may affect the held or waiting calls. The following are some of the
default short codes that can be used:
• *26: Clear CW Drop the previous call and answer the waiting call.
• *52: Clear Call Drop the previous call.
• *47: Conference Add Start a conference between the user and any held calls.
• Else, if there is a call waiting, that call is answered.
• Else, if there is a call on hold, that call is reconnected.
If the user presses Release or Drop or goes on-hook during a call, the current call is ended and
the user's phone returns to idle. If there is a waiting call, it starts ringing. The user can answer the
call by going off hook or pressing Hold.
With the phone idle:
If the user goes off hook:
• The first alerting call appearance is answered if any.
• Else, the first idle call appearance is seized and the user hears dial tone.
• The user can press Holdto switch between virtual call appearances. This will answer or
retrieve any call on next virtual call appearance or else hear dial tone to make a call.
With the phone idle but a call alerting:
Going off-hook or pressing Hold will answer the call.
When all the users virtual call appearances are in use, they are busy to any further calls. Calls will
follow forward on busy if set, else go to voicemail is available or else get busy indication.
The only other appearance button controls applied and supported are
Reserve Last CA This setting can be enabled for the extension user. When selected, the last
available call appearance is reserved for outgoing calls only. For example, for a user with 3 call
appearances, they return busy to any further calls when 2 virtual appearances are in use. The
extension user can press hold to get dial tone on the reserved call appearance. An available call
appearance is also required when using Feature 70 to initiate a call transfer.
Coverage Appearances Other users can have Coverage Appearance buttons set to provided
coverage to the virtual call appearance user. The virtual appearance users Individual Coverage
Time setting is applied.

Automatic Callback
Sets a ringback on the extension being called. When the target extension ends its current call, the
ringback user is rung (for their set No Answer Time) and if they answer, a new call is made to the
target extension.
Ringback can also be cleared using the Cancel Ring Back When Free function.

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 Auto-Intercom Deny

Details
• Action: Emulation | Automatic Callback.
• Action Data: None.
• Default Label: AutCB or Auto Callback.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Auto-Intercom Deny
Use the Auto-Intercom Deny function to block automatic intercom calls.
Details
• Action: Advanced | Do Not Disturb | Auto Intercom Deny.
• Action Data: Blank.
• Default Label: NoAI or No Auto Int Calls.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

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Button Programming Actions

- M-Series and T-Series.

Automatic Intercom
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Details
• Action: Emulation | Automatic Intercom.
• Action Data: User number or name. This field can be left blank for number entry when
pressed. On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.
• Default Label: Iauto or Auto Intercom.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Break Out
This feature is usable within a system multi-site network. It allows a user on one system in the
network to specify that the following dialing be processed by another system on the network as if
the user dialed it locally on that other system.
On phones with a multi-line display, if the target system is not specified in the button settings, a
menu of the available systems in the network is displayed from which a selection can be made.
Details
• Action: Advanced | Dial | Break Out.
• Action Data: Optional. The system name or IP address of the required system can be
specified. If no system name or IP address is set, on display phones a list of systems within
the network is displayed when the button is pressed.

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 Bridged Appearance

• Default Label: BkOut or Breakout.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

Bridged Appearance
Creates an appearance button that follows the state of another user's call appearance button. The
bridged appearance can be used to make and answer calls on behalf of the call appearance user.
For details, see Bridged Appearance Buttons on page 1166.
The bridged appearance button user must also have at least one call appearance button
programmed.
Bridged appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Appearance buttons can be set with a ring delay if required or to not ring. This does not affect
the visual alerting displayed next to the button. The delay uses the user's Ring Delay (User >
Telephony > Multi-line Options) setting.
Details
• Action: Appearance | Bridged Appearance.
• Action Data: User name and call appearance button number.
• Default Label: <user name><call appearance label>.
• Toggles: No.
• Status Indication: Yes. Required.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. Not supported on T7000, T7100, M7100 and M7100N.

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Button Programming Actions

Busy
Not used.

Busy On Held
When on, busy on held returns busy to new calls while the user has an existing call on hold. While
this feature can be used by users with appearance keys, it is not recommended as this overrides
the basic call handling intent of appearance keys.
Details
• Action: Advanced | Busy | Busy on Held.
• Action Data: 1 for on, 0 for off.
• Default Label: BusyH.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Call Forwarding All

Switches forward unconditional on and sets the forward number to the number specified or
prompts the user to enter a number if none is specified.
Details
• Action: Emulation | Call Forwarding All.
• Action Data: Telephone number or blank for entry when pressed.
- If blank, user's with a log in code will be prompted to enter that code to use this function.
- On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.
• Default Label: CFrwd or Call Forward All.
• Toggles: Yes.
• Status Indication: Yes.

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 Call Intrude

Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Call Intrude
This feature allows you to intrude on the existing connected call of the specified target user. All call
parties are put into a conference and can talk to and hear each other. A Call Intrude attempt to a
user who is idle becomes a Priority Call.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
• Users can use privacy features to set a call cannot be intruded on and recorded.
• Intruding onto a user doing silent monitoring (see Call Listen on page 960) is turned into a
silent monitoring call.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Action: Advanced | Call | Call Intrude.
• Action Data: User number or blank for entry when pressed. On large display phones, if
configured without a preset target, this type of button will display an interactive button menu
for target selection.
• Default Label: Intru or Intrude.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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Button Programming Actions

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Listen
This feature allows you to monitor another user's call without being heard.
• By default, monitoring is accompanied by a tone heard by all parties. Use of the tone is
controlled by the Beep on Listen setting on the System > Telephony > Tones and Music
tab.
Warning:
• Listening to a call without the other parties being aware is subject to local regulations.
You must ensure that you have complied with the local regulations. Failure to do so can
result in penalties.
The use of call listen is dependent on:
• The target being a member of the group set as the user's Monitor Group (User >
Telephony > Supervisor Settings). The user does not have to be a member of the group.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
A number of features are supported for call listening:
• Users can use privacy features to set a call cannot be intruded on and recorded.
• IP extensions can be monitored including those using direct media.
• The monitoring call can be initiated even if the target user is not currently on a call and
remains active until the monitoring user clears the monitoring call.
• The user who initiated the call listen can also record the call.
Intruding onto an a user doing silent monitoring (Call Listen) is turned into a silent monitoring call.
1400, 1600, 9500 and 9600 Series phones with a user button can initiate listening using that
button if the target user meets the criteria for listening.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Action: Advanced | Call | Call Listen.
• Action Data: User number.

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 Call Log

• Default Label: Listn or Listen.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Log
This function provides access to a list of received calls.
Details
• Action: Advanced | Call | Call Log.
• Action Data: None.
• Default Label: Call Log.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- M-Series and T-Series.

Call Park
Users can use a button set to this action to park and unpark calls.
• With a call connected, pressing the button will park that call.
• With no call connected, pressing the button displays call details and allows call retrieval.
The button can be configured either a specified park slot number or no specified park slot:
• When associated with a specific park slot number:
The button will park and unpark a call from that park slot, and indicate when there is a call is
parked in that park slot.

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Button Programming Actions

• When not associated with a specific park slot number:

The button can park up to 10 calls by assigning each a park slot number based on the user's
extension number. For example, for extension XXX, the first parked call is assigned to park
slot XXX0, the next to XXX1 and so on up to XXX9. The button will indicate when there are
parked calls in any of those slots.
Park button on other phones and in applications (for example IP Office SoftConsole and Avaya
one-X Portal) with the same park slot number as a parked call also indicate the park call and can
be used to retrieve it.
Details
• Action: Emulation > Call Park
• Action Data: Either blank or a specific park slot number.
- Park slot IDs can be up to 15 digits in length.
- Names can also be used for application park slots.
• Default Label: CPark or Call Park.
• Toggles: .
• Status Indication: .
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series, M-
Series
- Calls parked by Green flash Green flash Blue Slow flash
extension
- Call Parked by Red flash Red flash Green Slow flash
other extension
- No parked calls Off Off Gray Off
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 74.

Call Park and Page

Parks the user's current call into the park slot number specified on the System | Telephony | Park
& Page tab, in the Central Park Range field.
On M/T-series phones, 14xx/16xx phones, and the 9504 phone, the user is presented with up to
three Page Target Groups. On other 95xx/96xx phones, the Page action displays a scrolling list of

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 Call Park To Other Extension

possible Page Target Groups. The user may also directly enter a Page target number, or use the
system Directory to find a Page target.
A call Parked within the Central Park Range (regardless of the origin of the Park action) can be
retrieved by directly dialing the desired Central Park Range slot on which that call is Parked.
Details
• Action: Emulation | Call Park and Page.
• Action Data: None.
• Default Label: ParkPage
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. Feature 74 is equivalent to this button when a Central Park Range is defined. On
an M7000 phone, if this feature is invoked, the call always attempts to Park on the
highest defined Central Park Range slot. See the Call Park and Page short code
description for details.
- 1100 Series and 1200 Series.

Call Park To Other Extension

Allows the user to park their current call against another user's extension. The parked call
indication on that extension is then activated according to the telephone type.
If the target extension has a Call Park button with no specific park slot number, the parked call
will be indicated by that button and can be unparked from the list of parked calls shown when that
button is pressed.
The park slot number assigned to the parked call is based on the number of the extension parking
the call. For example, calls parked by extension 201 are assigned the park slot ID 2010, 2011 and
so on up to 2019 depending on the number of calls parked.
Details
• Action: Emulation | Call Park To Other Extension.
• Action Data: User number. This field can be left blank for number entry when pressed. On
large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.

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Button Programming Actions

• Default Label: RPark or Call Park to Other.

• Toggles: Yes .
• Status Indication: Yes. This is the status indication on the extension parking the call.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series, M-
Series
Parked call Green flash Green flash Blue Slow flash
No parked call Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Pickup
Answer an alerting call on the system.
Details
• Action: Emulation | Call Pickup.
• Action Data: None.
• Default Label: CpkUp or Call Pickup Any.
• Toggles: No.
• Status Indication: No.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Pickup Any

Pick up the first available ringing call on the system.

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 Call Pickup Group

Details
• Action: Advanced | Call | Call Pickup Any.
• Action Data: None.
• Default Label: PickA or Pickup Any.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Call Pickup Group

Pick up a call ringing any hunt group of which the user is a member or set to pick up calls from a
specific group.
The user can use this feature even if their membership of the group is currently set as disabled.
Details
• Action: Advanced | Call | Call Pickup Group.
• Action Data: Optional. To pick up calls from a specific group, use the group number or name.
• Default Label: PickG or Pickup Group.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 75.

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Button Programming Actions

Call Pickup Members

This feature can be used to pick up any call to an extension that is a member of the hunt group
specified. The call picked up does not have to be a hunt group call. The function includes group
members even if their membership of the group is currently disabled.
Details
• Action: Advanced | Call | Call Pickup Members.
• Action Data: Group number or name.
• Default Label: PickM or Pickup Members.
• Toggles: No.
• Status Indication: Yes (11.1 SP1)
- On suitable phones, pressing the button displays a list of any group member with a call
waiting to be answered. Pressing the button next to the user name answers their call.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Queue
Transfer the call to the target extension if free or busy. If busy, the call is queued to wait for the
phone to become free. This is similar to transfer except it allows you to transfer calls to a busy
phone.
Details
• Action: Advanced | Call | Call Queue.
• Action Data: User number.
• Default Label: Queue.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.

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 Call Record

- 1400 Series and 1600 Series.

- M-Series and T-Series.
- 1100 Series and 1200 Series.

Call Record
This feature allows you to record a conversation and requires Voicemail Pro to be installed.
• An advice of recording warning will be given if configured on the voicemail system.
• The recording is placed in the mailbox specified by the user's Manual Recording Mailbox
setting.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
• Users can use privacy features to set a call cannot be intruded on and recorded.
Details
• Action: Advanced | Call | Call Record.
• Action Data: None.
• Default Label: Recor or Record.
• Toggles: Yes.
• Status Indication: Yes.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.
- This button action is also supported by the Vantage Connect Expansion application.

Call Screening
This function is used to enable or disable call screening. While enabled, when a caller is
presented to the user's voicemail mailbox, if the user's phone is idle they will hear through the
phone's handsfree speaker the caller leaving the message and can select to answer or ignore the
call.

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Button Programming Actions

This feature can be used with both Embedded Voicemail and Voicemail Pro. Call screening is only
applied as follows:
• It is only applied to calls that have audible alerted at the user's extension before going to
voicemail. This requires the user to have both voicemail coverage and call screening enabled
and the phone's ringer not set to silent. However it is not applied if the user transfers the call
to voicemail.
• It is only applied if the user's phone is idle. That is, not on a call or with a call held pending
transfer or conference.
• Calls that ring the user, are then rerouted (for example follow a forward on busy setting) and
then return to the user's mailbox are screened.
While a call is being screened, the phone can be used to either answer or ignore the screened
call. Auto answer options are ignored.
Answering a screened call
A screened call can be answered by pressing the Answer soft key (if displayed) or lifting the
handset. Pressing the call appearance or line button on which the call is indicated will also answer
the call.
When answered:
• The phone's microphone is unmuted and a normal call between the user and caller now
exists.
• The voicemail recording stops but that portion of the call already recorded is left as a new
message in the user's mailbox.
Ignoring a screened call
A screened call can be ignored by pressing the Ignore soft key if displayed. On 1400, 1600, 9500
and 9600 Series phones, pressing the SPEAKER button will ignore the call. On M-Series and
T-Series phones, pressing the Release key will ignore the call.
When ignored:
• The call continues to be recorded until the caller hangs up or transfers out of the mailbox.
• The user's phone returns to idle with call screening still enabled. However any other call that
has already gone to voicemail is not screened.
Screened call operation
While a call is being screened:
• The mailbox greeting played and the caller can be heard on the phone's speakerphone. The
caller cannot hear the user.
• The user is regarded as being active on a call. They will not be presented with hunt group
calls and additional personal calls use abbreviated ringing.
• 1400/1600/9500/9600 Series phones: If the phone's default audio path is set to headset or
the phone is idle on headset, then the screened call is heard through the headset.
• Any additional calls that go to the user's mailbox when they are already screening a call,
remain at the mailbox and are not screened even if the existing call being screened is ended.
• Making or answering another call while listening to a screened call is treated as ignoring
the screened call. For users with Answer Pre-Select enabled (User | Telephony | Multi-line

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 Call Screening

Options), pressing an appearance button to display details of a call is also treated as ignoring
the screened call.
• Other users cannot access a call that is being screened. For example they cannot use call
pickup, bridged appearance or line appearance buttons, call intrude or call acquire functions.
• Phone based administration cannot be accessed and the hold, transfer and conference
buttons are ignored.
• The screened caller using DTMF breakout ends the call screening.
Enabling do not disturb overrides call screening except for calls from numbers in the user's do not
disturb exceptions list.
Locking the phone overrides call screening.
Manual call recording cannot be applied to a call being screened.
While a call is being screened, it uses one of the available voicemail channels. If no voicemail
channels are available, call screening does not occur.
Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
Details
• Action: Advanced | Call | Call Screening.
• Action Data: None.
• Default Label: CallScreen or Call Screening.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. Not T7406E.

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Button Programming Actions

Call Steal
This function allows a user to seize a call answered or ringing on another extension. This function
can be used with or without a specified user target.
• If the target has multiple alerting calls, the function steals the longest waiting call.
• If the target has a connected call and no altering calls, the function steals the connected call.
This is subject to the Can Intrude setting of the Call Steal user and the Cannot Be Intruded
setting of the target.
• If no target is specified, the function attempts to reclaim the user's last ringing or transferred
call if it has not been answered or gone to voicemail.
• Stealing a video call changes the call to an audio call.
• R11.1 FP2 SP4 and higher: The shortcode for this feature can be used with the user's
own extension number. That enables twinned and simultaneous device users to move a
connected call from another one of their devices. This usage ignores the user's privacy and
intrusion settings.
Details
• Action: Advanced | Call | Call Steal.
• Action Data:
- User number or blank for last call transferred.
• Default Label: Aquir or Aquire.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Call Waiting Off

Switches call waiting off for the user. This button function is obsolete. The Call Waiting On button
function toggles on/off and indicates current status.
Details
• Action: Advanced | Call | Call Waiting Off.
• Action Data: None.

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 Call Waiting On

• Default Label: CWOff.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Call Waiting On
Enables call waiting on the user's extension. When the user is on a call and another call arrives,
they will hear a call waiting tone.
Note:
Call waiting does not operate for user's with call appearance buttons. See Call Waiting.
Details
• Action: Advanced | Call | Call Waiting On.
• Action Data: None.
• Default Label: CWOn or Call Waiting On.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Call Waiting Suspend

Disables call waiting, if on, for the duration of the extension's next call.
Details
• Action: Advanced | Call | Call Waiting Suspend.

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Button Programming Actions

• Action Data: None.

• Default Label: CWSus.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Cancel All Forwarding

Cancels forward unconditional, forward on busy, forward on no answer, follow me and do not
disturb if any of those are active on the user's extension.
• Action: Advanced | Call | Cancel All Forwarding.
• Action Data: None.
• Default Label: FwdOf or Call Forward Off.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
Details
• 9500 Series, 9600 Series and J100 Series.
• 1400 Series and 1600 Series.
• M-Series and T-Series.
• This button action is also supported by the Vantage Connect Expansion application.

Cancel Leave Word Calling

Supported for CTI emulation only.
Cancels the last Leave Word Calling message originated by the user.
Details
• Action: Emulation | Cancel Leave Word Calling.

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 Cancel Ring Back When Free

• Action Data: None.

• Default Label: CnLWC.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Cancel Ring Back When Free

Cancels any existing ring back set by the user, see Ring Back When Free. Note that the Ring
Back When Free button toggles to set or cancel ring back when free and also indicates the current
status.
Details
• Action: Advanced | Miscellaneous | Cancel Ring Back When Free.
• Action Data: None.
• Default Label: RBak-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
1. M-Series/T-Series: The button is equivalent to Feature #2.

Channel Monitor
For Avaya use only. Configurable through web manager only.

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Button Programming Actions

Clear Call
This feature can be used to end the last call put on hold. This can be used in scenarios where a
first call is already on hold and simply ending the second call will cause an unsupervised transfer
of the first call.
Details
• Action: Advanced | Call | Clear Call.
• Action Data: None.
• Default Label: Clear.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Clear CW
End the user's current call and answer any call waiting. Requires the user to also have call waiting
indication on. This function does not work for users with multiple call appearance buttons.
Details
• Action: Advanced | Call | Clear CW.
• Action Data: None.
• Default Label: ClrCW.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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 Clear Hunt Group Night Service

Clear Hunt Group Night Service

Changes the specified hunt group from Night Service mode to 'In Service' mode. This button
function is obsolete. The Set Hunt Group Night Service function can be used to toggle a group
in/out of service and provides lamp status indication.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Details
• Action: Advanced | Call | Clear Hunt Group Night Service.
• Action Data: Group number. If left blank, the button will affect all hunt groups of which the
user is a member.
- The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively. The
service is indicated by setting the service name as the telephone number or action data.
Do not use quotation marks.
• Default Label: HGNS-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Clear Hunt Group Out Of Service

Changes the specified hunt groups status from Out of Service mode to 'In Service' mode. This
button function is obsolete. The Set Hunt Group Out Of Service function can be used to toggle a
group in/out of service and provides lamp status indication.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Details
• Action: Advanced | Call | Clear Hunt Group Out of Service.
• Action Data: Group number. If left blank, the button will affect all hunt groups of which the
user is a member.

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Button Programming Actions

• Default Label: HGOS-.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Clear Quota
Quotas can be assigned on outgoing calls to data services such as internet connections. The
quota defines the number of minutes available for the service within a time frame set within the
service, for example each day, each week or each month.
The Clear Quota function can be used to reset the quota for a specific service or for all services.
Details
• Action: Advanced | Call | Clear Quota.
• Action Data: Service name" or "" (all services).
• Default Label: Quota.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Coaching Intrusion
This feature allows the you to intrude on another user's call and to talk to them without being
heard by the other call parties to which they can still talk. For example: User A is on a call with
user B. When user C intrudes on user A, they can hear users A and B but can only be heard by
user A.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.

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 Conference

• Listening to a call without the other parties being aware is subject to local regulations. You
must ensure that you have complied with the local regulations. Failure to do so can result in
penalties.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Action: Advanced | Call | Coaching Intrusion.
• Action Data: User number or name or blank for entry when pressed.
• Default Label: Coach or Coaching Intrusion.
• Toggles: No.
• Status Indication: No.
• User Admin: No feedback provided..
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- Not supported on non-IP telephones when using a headset.

Conference
This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same conference process as dialing Feature 3.
Details
• Action: Advanced | Call | Conference.
• Action Data: None.
• Default Label: Conf or Conference Add.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- M-Series and T-Series.
• The button is equivalent to Feature 3.

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Button Programming Actions

Conference Add
Conference add controls can be used to place the user, their current call and any calls they have
on hold into a conference. When used to start a new conference, the system automatically assigns
a conference ID to the call. This is termed ad-hoc (impromptu) conferencing.
If the call on hold is an existing conference, the user and any current call are added to that
conference. This can be used to add additional calls to an ad-hoc conference or to a meet-me
conference. Conference add can be used to connect two parties together. After creating the
conference, the user can drop from the conference and the two incoming calls remain connected.
For R11.0 and higher, the button has additional features:
• When pressed during a normal two-party call, that call is turned into a two-party conference
call. This then provides access to the phone’s other conference control, such as to add other
parties, without interrupting the call.
• During an existing conference, pressing the button (on 1400, 1600, 9500, 9600 and J100
Series phones) provides a menu to enter the number of an additional party to add to the
conference without put the conference on hold. The other parties in the conference can hear
the call progress and if answered the other party is immediately in the conference.
For further details, see Conferencing on page 913.
Details
• Action: Advanced | Call | Conference Add.
• Action Data: None.
• Default Label: Conf+ or Conference Add.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Conference Meet Me
Conference meet-me refers to features that allow a user or caller to join a specific conference by
using the conference's ID number (either preset in the button's configuration or entered at the time
of joining the conference).

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 Conference Meet Me

Note:
• Conference Meet Me features can create conferences that include only one or two
parties. These are still conferences that are using resources from the host system's
conference capacity.
Conference ID Numbers
Each conference has a conference ID number:
• Ad-Hoc Conferences - By default, ad-hoc conferences are assigned numbers starting from
100 for the first conference in progress. Therefore, for conference Meet-Me features, you
should always specify a number away from this range ensure that the conference joined is
not an ad-hoc conference started by other users. It is not possible to join a conference using
conference Meet-Me features when the conference ID is in use by an ad-hoc conference.
• User Personal Meet-Me Conferences - Each user's own extension number is treated as
their own personal conference number. Only that user is able to start a conference using
that number as the conference ID. Any one else attempting to start a conference with that
number will find themselves in a conference but on hold until the owner also joins. Personal
conferences are always hosted on the owner's system.
• System Meet-Me Conferences - Each of these is assigned a conference ID number when
the conference settings are configured.
For further details, see Conferencing on page 913.
Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE 18 service.
Multi-Site Network Conferencing
Meet Me conference IDs are now shared across a multi-site network. For example, if a conference
with the ID 500 is started on one system, anyone else joining conference 500 on any system will
join the same conference. Each conference still uses the conference resources of the system on
which it was started and is limited by the available conference capacity of that system.
Previously separate conferences, each with the same conference ID, could be started on each
system in a multi-site network.
Other Features
• Transfer to a Conference Button - A currently connected caller can be transferred into the
conference by pressing TRANSFER, then the Conference Meet Me button and TRANSFER
again to complete the transfer. This allows the user to place callers into the conference
specified by the button without being part of the conference call themselves. This option is
only support on Avaya phones with a fixed TRANSFER button.
• Conference Button Status Indication - When the conference is active, any buttons
associated with the conference ID indicate the active state.
Details
• Action: Advanced | Call | Conference Meet Me.

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Button Programming Actions

• Action Data: Conference number. This can be an alphanumeric value up to 15 characters.

- User Personal Conference Number Each user's own extension number is treated as
their own personal conference number. Only that user is able to start a conference using
that number as the conference ID. Any one else attempting to start a conference with
that number will find themselves in a conference but on hold until the owner also joins.
Personal conferences are always hosted on the owner's system.
- When a user calls from their mobile twinned number, the personal conference feature will
only work if they access the conference using an FNE18 service.
• Default Label: CnfMM <conference number> or Conf. Meet Me <conference number>.
• Toggles: No.
• Status Indication: Yes
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

For a Conference Meet Me configured to the user's own extension number, the indicator
flashes red when the conference is in use but the user has not joined. There is also an
abbreviated ring when the indicator changes to flashing red. It changes to solid red when the
user joins.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Consult
Supported for CTI emulation only.
Details
• Action: Emulation | Consult.
• Action Data: None.
• Default Label: Cnslt.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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 Coverage Appearance

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Coverage Appearance
Creates a button that alerts when a call to the specified covered user is unanswered after that
users Individual Coverage Timer expires. For details, see Call Coverage Buttons on page 1171.
The call coverage appearance button user must also have at least one call appearance button
programmed. The covered user does not need to be using call appearance buttons.
Coverage appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Appearance buttons can be set with a ring delay if required or to not ring. This does not affect
the visual alerting displayed next to the button. The delay uses the user's Ring Delay (User >
Telephony > Multi-line Options) setting.
Details
• Action: Appearance | Coverage Appearance.
• Action Data: User name.
• Default Label: <user name>.
• Toggles: No.
• Status Indication: Yes.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial
This action is used to dial the number contained in the Telephone Number field. A partial number
can be enter for the user to complete. On buttons with a text label area, Dial followed by the
number is shown.
Details
• Action Data: Telephone number or partial telephone number.

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Button Programming Actions

• Default Label: Dial.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- This button action is also supported by the Vantage Connect Expansion application.
• 9500 Series, 9600 Series and J100 Series.
• 1400 Series and 1600 Series.
• M-Series and T-Series.

Dial 3K1
The call is presented to local exchange as a "3K1 Speech Call". Useful in some where voice calls
cost less than data calls.
Details
• Action: Advanced | Dial | Dial 3K1.
• Action Data: Telephone number.
• Default Label: D3K1 or Dial 3K1.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial 56K
The call presented to local exchange as a "Data Call".
Details
• Action: Advanced | Dial | Dial 56K.

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 Dial 64K

• Action Data: Telephone number.

• Default Label: D56K or Dial 56K.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial 64K
The call is presented to local exchange as a "Data Call".
Details
• Action: Advanced | Dial | Dial 64K.
• Action Data: Telephone number.
• Default Label: D64K or Dial 64K.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial CW
Call the specified extension number and force call waiting indication on if the extension is
already on a call. The call waiting indication will not work if the extension called has multiple
call appearance buttons in use.
Details
• Action: Advanced | Dial | Dial CW.

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Button Programming Actions

• Action Data: User number.

• Default Label: DCW or Dial Call Waiting.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Direct
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Details
• Action: Advanced | Dial | Dial Direct.
• Action Data: User number or name or blank for entry when pressed. If left blank, the Dial
Direct button can be used with User buttons to specify the target.
• Default Label: Dirct or Auto Intercom.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

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 Dial Emergency

Dial Emergency
Dials the number specified regardless of any outgoing call barring applicable to the user. See
Configuration for Emergency Calls on page 652.
• Details of calls made using this function can be viewed using an Emergency View button.
See Emergency View on page 1103.
Details
• Action: Advanced | Dial | Dial Emergency.
• Action Data: Telephone number. This must match the emergency call routing configured for
the system or for the extension location.
• Default Label: Emrgy or Dial Emergency.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Inclusion
This feature allows you to intrude on another user's call to talk to them. Their current call is put
on hold while you talk and automatically reconnected when you end the intrusion. The intruder
and the target extension can then talk but cannot be heard by the other party. This can include
intruding into a conference call, where the conference will continue without the intrusion target.
During the intrusion all parties hear a repeated intrusion tone. When the intruder hangs-up the
original call parties are reconnected. Attempting to hold a dial inclusion call simply ends the
intrusion. The inclusion cannot be parked.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Action: Advanced | Dial | Dial Inclusion.

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Button Programming Actions

• Action Data: User number or name or blank for user selection when pressed. On large
display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
• Default Label: Inclu or Dial Inclusion.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Intercom
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Details
• Action: Emulation | Dial Intercom.
• Action Data: User number or name or blank for number entry when pressed. On large
display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
• Default Label: Idial or Auto Intercom.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 66 <number>.

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 Dial Paging

Dial Paging
Makes a paging call to an extension or group specified. If no number is specified, this can be
dialed after pressing the button. The target extension or group members must be free and must
support handsfree auto-answer in order to hear the page.
On Avaya phones with a CONFERENCE button, a paged user can convert the page call into a
normal call by pressing that button.
Details
• Action: Advanced | Dial | Dial Paging.
• Action Data: User number or name or group number or name or blank for number entry
when pressed.
• Default Label: Page.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- This button action is also supported by the Vantage Connect Expansion application.
• 9500 Series, 9600 Series and J100 Series.
• 1400 Series and 1600 Series.
• M-Series and T-Series.
• 1100 Series and 1200 Series.

Dial Physical Extn by Number

Call the specified extension using its Base Extension number setting. This is regardless of the
current user logged in at that extension and any forwarding, follow me or do not disturb settings
applied by the extension user. This function requires the extension to be assigned a default
extension number in the system configuration. If the extension does not have a default extension
number, Dial Physical Extn by Id should be used.
Details
• Action: Advanced | Dial | Dial Physical Extn by Number.
• Action Data: Extension port base extension number.
• Default Label: PhyEx or Dial Physical Extn.
• Toggles: No.
• Status Indication: No.

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Button Programming Actions

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Physical Number by ID

Call the specified extension, if free, regardless of the current user logged in at that extension and
any forwarding, follow me or do not disturb settings applied by the extension user. This function
uses the port ID shown in the system configuration.
Details
• Action: Advanced | Dial | Dial Physical Extn by Id.
• Action Data: Extension port ID number.
• Default Label: DialP or Dial Extn by Id.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Speech
This feature allows a short code to be created to force the outgoing call to use the Speech bearer
capability.
Details
• Action: Advanced | Dial | Dial Speech.
• Action Data: Telephone number.
• Default Label: DSpch or Dial Speech.
• Toggles: No.

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 Dial V110

• Status Indication: No.

• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial V110
The call is presented to local exchange as a "Data Call".
Details
• Action: Advanced | Dial | Dial V110.
• Action Data: Telephone number.
• Default Label: DV110 or Dial V110.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial V120
The call is presented to local exchange as a "Data Call".
Details
• Action: Advanced | Dial | Dial V120.
• Action Data: Telephone number.
• Default Label: DV120 or Dial V120.
• Toggles: No.
• Status Indication: No.

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Button Programming Actions

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Dial Video
The call is presented to the local exchange as a "Video Call".
Details
• Action: Advanced | Dial | Dial Video.
• Action Data: Telephone number.
• Default Label: Dvide or Dial Video.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Directed Call Pickup

Pickup a call ringing at a specific extension or hunt group.
Details
• Action: Emulation | Directed Pickup.
• Action Data: User number or name or group number or name or blank for number entry
when pressed. On large display phones, if configured without a preset target, this type of
button will display an interactive button menu for target selection.
• Default Label: DpkUp or Call Pickup.
• Toggles: No.
• Status Indication: No.

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 Directory

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 76.
- 1100 Series and 1200 Series.

Directory
A Dir button provides access to various directories and allows telephone number selection by
dialed name matching. The directories available for searching depend on the phone type, see
User Directory Access. Once they user has selected a directory, dialing on the dial pad letter keys
is used to filter the display of matching names, with controls for scrolling through the matching
names and for calling the currently displayed name.
Details
• Action: Emulation | Directory.
• Action Data: None.
• Default Label: Dir.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Display Msg
Allows the sending of text messages to digital phones on the local system.
Details
• Action: Advanced | Dial | Display Msg.

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Button Programming Actions

• Action Data: The telephone number takes the format N";T" where:
- N is the target extension.
- T is the text message. Note that the "; before the text and the " after the text are required.
• Default Label: Displ.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Do Not Disturb Exception Add

Adds a number to the user's "Do Not Disturb Exception List". This can be the number of an
internal user or a number to match the CLI of a particular external caller. Calls from that number,
except hunt group calls, will ignore the user's Do Not Disturb setting. For further details see Do
Not Disturb (DND).
Details
• Action: Advanced | Do Not Disturb | Do Not Disturb Exception Add.
• Action Data: Telephone number or CLI. Up to 31 characters. For CLI numbers any prefix
added by the system must also be included.
• Default Label: DNDX+.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

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 Do Not Disturb Exception Delete

Do Not Disturb Exception Delete

Removes a number from the user's "Do Not Disturb Exception List". This can be the number of an
internal user or a number to match the CLI of a particular external caller.
Details
• Action: Advanced | Do Not Disturb | Do Not Disturb Exception Delete.
• Action Data: Telephone number or CLI.
• Default Label: DNDX-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Do Not Disturb Off

Cancels the user's 'do not disturb' mode if set. This button function is obsolete as the do not
disturb on function toggles on/off and indicates the button status.
Details
• Action: Advanced | Do Not Disturb | Do Not Disturb Off.
• Action Data: None.
• Default Label: DNDOf.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- 1100 Series and 1200 Series.

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Button Programming Actions

Do Not Disturb On
Enables the user's 'do not disturb' mode.
Details
• Action: Advanced | Do Not Disturb | Do Not Disturb On.
• Action Data: None.
• Default Label: DNDOn or Do Not Disturb.
• Toggles: Yes.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 85.
- 1100 Series and 1200 Series.
- This button action is also supported by the Vantage Connect Expansion application.

Drop
This action is supported on phones which do not have a permanent Drop button.
• For a currently connected call, pressing Drop disconnects the call. When drop is used to
end a call, silence is returned to the user rather than dial tone. This is intended operation,
reflecting that Drop is mainly intended for use by call center headset users.
• If the user has no currently connected call, pressing Drop will redirect a ringing call using the
user's Forward on No Answer setting if set or otherwise to voicemail if available.
• For a conference call, on phones with a suitable display, Drop can be used to display the
conference parties and allow selection of which party to drop from the conference.
Details
• Action: Emulation | Drop.
• Action Data: None.
• Default Label: Drop or Drop Call.
• Toggles: No.
• Status Indication: No.

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 Emergency View

• User Admin: .
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.

Emergency View
A button set to this function indicates when a call has been made from the system to which the
user’s extension is registered. The definition of an emergency call is one using a number routed
by a Dial Emergency button or short code.
• Pressing the button displays details of currently connected emergency calls (the first 10).
• After pressing the button, the History option displays details of any previously connected
emergency calls (the first 30) and allows deletion those call details.
• The emergency call history for a system is shared by all users on the same system.
Therefore updates to or deleting the history affects the details shown on all user phones
on the same system.
• The time shown in the call details, is the UTC time of the alarm calls. On J189 phones, it also
includes the location name if an IP Office Location entry was used to route the call.
• Note that the button only works for an extension registered to the same system as the
outgoing trunk used for the emergency call.
Details
• Action: Emulation | Emergency View.
• Action Data: None
• Default Label: 911–View or EView
• Toggles: No.
• Status Indication: Yes
- The button gives a single ring and then flashes when there is a connected emergency call
in progress.
- The button remains on when there are previous emergency calls in the alarm history.
- Note that there is a delay of a few seconds in changes of the lamp state.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

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Button Programming Actions

Extn Login
Extn Login allows a user who has been configured with a Login Code (User | Telephony |
Supervisor Settings) to take over ownership of any extension. That user's extension number
becomes the extension number of the extension while they are logged. This is also called ‘hot
desking’.
Hot desking is not supported for H175, E129 and J129 telephones.
When used, the user is prompted to enter their extension number and then their log in code. Login
codes of up to 15 digits are supported with Extn Login buttons. Login codes of up to 31 digits are
supported with Extn Login short codes.
When a user logs in, as many of their user settings as possible are applied to the extension. The
range of settings applied depends on the phone type and on the system configuration.
By default, on 1400 Series, 1600 Series, 9500 Series and 9600 Series phones, the user's call log
and personal directory are accessible while they are logged in. This also applied to M-Series and
T-Series telephones.
On other types of phone, those items such as call logs and speed dials are typically stored locally
by the phone and will not change when users log in and log out.
If the user logging in was already logged in or associated with another phone, they will be
automatically logged out that phone.
Details
• Action: Advanced | Extension | Extn Login.
• Action Data: None.
• Default Label: Login.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

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 Extn Logout

Extn Logout
Logs out a user from the phone. The phone will return to its normal default user, if an extension
number is set against the physical extension settings in the configuration. Otherwise it takes the
setting of the NoUser user. This action is obsolete as Extn Login can be used to log out an
existing logged in user.
• If the user who logged out was the default user for an extension, dialing *36 will associate the
extension with the user unless they are set to forced log in.
• This feature cannot be used by a user who does not have a log in code.
Details
• Action: Advanced | Extension | Extn Logout.
• Action Data: None.
• Default Label: Logof or Logout.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.
- This button action is also supported by the Vantage Connect Expansion application.

Flash Hook
Sends a hook flash signal to the currently connected line if that line is an analog line.
Details
• Action: Advanced | Miscellaneous | Flash Hook.
• Action Data: Optional. Normally this field is left blank. It can contain the destination number
for a Centrex Transfer for external calls on a local analog line from a Centrex service
provider. See Centrex Transfer on page 791.
• Default Label: Flash or Flash Hook.
• Toggles: No.
• Status Indication: No.

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Button Programming Actions

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Follow Me Here
Causes calls to the extension number specified, to be redirected to this user's extension. User's
with a log in code will be prompted to enter that code when using this function. For further details,
see Follow Me on page 747.
Details
• Action: Advanced | Follow Me | Follow Me Here.
• Action Data: User name or user number.
- If a user name or user number has been entered in the Action Data field, when the
interactive menu opens, press Enter to activate Follow Me Here for the number displayed
on the screen.
- This field can be left blank for number entry when pressed.
- On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.
• Default Label: Here+ or Follow Me Here.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

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 Follow Me Here Cancel

Follow Me Here Cancel

Cancels any 'Follow Me Here' set on the specified extension. Only works if entered at the
extension to which the extension's calls are being sent by the follow me action. For further details,
see Follow Me on page 747.
Details
• Action: Advanced | Follow Me | Follow Me Here Cancel.
• Action Data: User number or blank for number entry when pressed.
- If a user name or user number has been entered in the Action Data field, when the
interactive menu opens, press Enter to deactivate Follow Me Here for the number
displayed on the screen.
- On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.
• Default Label: Here- or Follow Me Here-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Follow Me To
Leaving the extension blank prompts the user to enter the extension to which their calls should be
redirected. User's with a log in code will be prompted to enter that code when using this function.
For further details, see Follow Me on page 747.
Details
• Action: Advanced | Follow Me | Follow Me To.
• Action Data: User name or user number or blank for number entry when pressed.
- If a user name or user number has been entered in the Action Data field, when the
interactive menu opens, press Enter to activate Follow Me To for the number displayed
on the screen.
- On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.

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Button Programming Actions

• Default Label: FolTo or Follow Me To.

• Toggles: Yes.
• Status Indication: Yes. On/off status indication is provided if the button is programmed with
a user name or number.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Forward Hunt Group Calls Off

Cancels the forwarding of the user's hunt group calls. This function is obsolete since the button
function Forward Hunt Group Calls On toggles on/off and indicates status.
Details
• Action: Advanced | Forward | Forward Hunt Group Calls Off.
• Action Data: None.
• Default Label: FwdH-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Forward Hunt Group Calls On

Forward the user's hunt group calls (internal and external). This function only works when forward
unconditional is also on and uses the same forwarding number as forward unconditional.

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 Forward Number

This option is only applied for calls to Sequential and Rotary type hunt groups. Calls from other
hunt group types are not presented to the user when they have Forward Unconditional active.
Note also that hunt group calls cannot be forwarded to another hunt group.
Details
• Action: Advanced | Forward | Forward Hunt Group Calls On.
• Action Data: None.
• Default Label: FwdH+ or Fwd HG Calls.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Forward Number
Sets the number to which calls are forwarded when the user has forwarding on. Used for all
forwarding options unless a separate Forward On Busy Number is also set. Forwarding to
an external number is blocked if Inhibit Off-Switch Transfers is selected within the system
configuration.
Details
• Action: Advanced | Forward | Forward Number.
• Action Data: Telephone number.
• The field to be left blank to prompt the user for entry when the button is pressed. If blank,
users with a log in code will be prompted to enter that code.
• On large display phones, if configured without a preset target, this type of button will display
an interactive button menu for target selection.
• Default Label: FwdNo or Fwd Number.
• Toggles: No.
• Status Indication: Yes. For a button with a prefixed number, status indication will indicate
when that number matches the users current set number. For a button with a no number,
status indication will show when a number has been set.

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Button Programming Actions

Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Forward On Busy Number

Sets the number to which calls are forwarded when using 'Forward on Busy' and/or 'Forward
on No Answer'. Forwarding to an external number is blocked if Inhibit Off-Switch Transfers is
selected within the system configuration.
For further details, see Forward on Busy on page 751.
Details
• Action: Advanced | Forward | Forward on Busy Number.
• Action Data: Telephone number.
- The field to be left blank to prompt the user for entry when the button is pressed. If blank,
users with a log in code will be prompted to enter that code.
- On large display phones, if configured without a preset target, this type of button will
display an interactive button menu for target selection.
• Default Label: FwBNo or Fwd Busy Number.
• Toggles: No.
• Status Indication: Yes. For a button with a prefixed number, status indication indicates when
that number matches the user's current set number. For a button with a no number, status
indication shows when a number has been set.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

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 Forward On Busy Off

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Forward On Busy Off

Switches forward on busy off. This button function is obsolete, as Forward On Busy On can be
used to switch forward on busy on/off and provides status indication.
Details
• Action: Advanced | Forward | Forward on Busy Off.
• Action Data: None.
• Default Label: FwBOf.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Forward On Busy On
Enables forwarding when the user's extension is busy. For users with call appearance buttons,
they will only return busy when all call appearance buttons are in use. Uses the Forward Number
as its destination unless a separate Forward on Busy Number is set. For further details, see
Forward on Busy on page 751.
Details
• Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
• Action: Advanced | Forward | Forward on Busy On.
• Action Data: None.
• Default Label: FwBOn or Fwd Busy.
• Toggles: Yes.

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Button Programming Actions

• Status Indication: Yes.

Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Forward On No Answer Off

Switches forward on no answer off. This button function is obsolete, as Forward On No Answer
On can be used to switch forward on no answer on/off and provides status indication.
Details
• Action: Advanced | Forward | Forward on No Answer Off.
• Action Data: None.
• Default Label: FwNOf.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Forward On No Answer On
Switches forward on no answer on/off. The time used to determine the call as unanswered is the
user's no answer time. Uses the Forward Number as its destination unless a separate Forward
on Busy Number is set.
For further details, see Forward on No Answer on page 753.

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 Forward Unconditional Off

Details
• Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
• Action: Advanced | Forward | Forward on No Answer On.
• Action Data: None.
• Default Label: FwNOn or Fwd No Answer.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Forward Unconditional Off

Switch 'forward all calls' off. This does not affect 'Forward on No Answer' and/or 'Forward on Busy'
if also on. This function is obsolete as a button set to Forward Unconditional On toggles on/off and
indicates when on.
Details
• Action: Advanced | Forward | Forward Unconditional Off.
• Action Data: None.
• Default Label: FwUOf.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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Button Programming Actions

Forward Unconditional On
This function is also known as 'divert all' and 'forward all'. It forwards all calls, except hunt group
and page calls, to the forward number set for the user's extension. To also forward hunt group
calls to the same number 'Forward Hunt Group Calls On' must also be used.
For further details, see Forward Unconditional on page 749.
Details
• Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
- In addition to the lamp indication shown below, some phones display D when forward
unconditional is on.
• Action: Advanced | Forward | Forward Unconditional On.
• Action Data: None.
• Default Label: FwUOn or Fwd Unconditional.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 4 <number>.
- This button action is also supported by the Vantage Connect Expansion application.

Group
Monitors the status of a hunt group queue. This option is only supported for hunt groups with
queuing enabled. The user does not have to be a member of the group.
Depending on the users button type, indication is given for when the group has alerting calls and
queued calls (queued in this case is defined as more calls waiting than there are available group
members).
Pressing a Group button answers the longest waiting call.

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 Group Listen On

The definition of queued calls include group calls that are ringing. However, for operation of the
Group button, ringing calls are separate from other queued calls.
Details
• Action: Group.
• Action Data: Group name enclosed in " " double-quotes or group number.
• Default Label: <group name>.
• Toggles: No.
• Status Indication: Required.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series, M-
Series
- No calls Off Off Grey Off
- Call alerting Green flash Green flash Blue Slow flash
- Calls queued Red flash Red flash Green Slow flash

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Group Listen On
Using group listen allows callers to be heard through the phone's handsfree speaker but to only
hear the phone's handset microphone. When group listen is enabled, it modifies the handsfree
functionality of the user’s phone in the following manner
• When the user’s phone is placed in handsfree/speaker mode, the speech path from the
connected party is broadcast on the phone speaker but the phone's base microphone is
disabled.
• The connected party can only hear speech delivered through the phone's handset
microphone.
• Group listen is not supported for IP phones or when using a phone's HEADSET button.
• For T-Series and M- Series phones, this option can be turned on or off during a call. For other
phones, currently connected calls are not affected by changes to this setting, instead group
listen must be selected before the call is connected.
Group listen is automatically turned off when the call is ended.

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Button Programming Actions

Details
• Action: Advanced | Extension | Group Listen On.
• Action Data: None.
• Default Label: Group Listen On.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 9500 T-Series,
On. Green on On
Off. Off Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series, 9500 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 802 (On) and Feature #802 (Off).

Group Paging
Makes a paging call to an extension or group specified. If no number is specified, this can be
dialed after pressing the button. The target extension or group members must be free and must
support handsfree auto-answer in order to hear the page.
On Avaya phones, a paged user can convert the page call into a normal call by pressing the
Conference button.
Details
• Action: Emulation | Group Paging.
• Action Data: User number or name or group number or name. On large display phones, if
configured without a preset target, this type of button will display an interactive button menu
for target selection.
• Default Label: GrpPg.
• Toggles: No.
• Status Indication: Yes.
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.

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 Headset Toggle

- 1400 Series and 1600 Series.

- M-Series and T-Series.
1. The button is equivalent to Feature 60 <number>.
- 1100 Series and 1200 Series.

Headset Toggle
This function is intended for use with Avaya phones that have separate handset and headset
sockets but do not provide a dedicated Headset button. On phones without a headset socket or
with a dedicated headset button this control will have no effect.
Details
• Action: Miscellaneous | Headset Toggle.
• Action Data: None.
• Default Label: HdSet.
• Toggles: Yes.
• Status Indication: Yes.
• User Admin: No.

Hold Call
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The Hold Call feature "holds" the current call to a slot. The current call is
always automatically placed into slot 0 if it has not been placed in a specified slot. Only available if
supported by the ISDN exchange.
Details
• Action: Advanced | Hold | Hold Call.
• Action Data: ISDN Exchange hold slot number or blank (slot 0).
• Default Label: Hold.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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Button Programming Actions

Hold CW
Place the user's current call on hold and answers the waiting call. This function is not supported
on phones which have multiple call appearance buttons set.
Details
• Action: Advanced | Hold | Hold CW.
• Action Data: None.
• Default Label: HoldCW.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Hold Music
This feature allows the user to listen to the system's music on hold. See Music On Hold for more
information.
Details
• Action: Advanced | Hold | Hold Music.
• Action Data: Optional. Systems can support multiple hold music sources. However only the
system source is supported for Hold Music buttons.
• Default Label: Music or Hold Music.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

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 Hunt Group Enable

Hunt Group Enable

An individual users membership of any particular hunt groups is programmed through the system
configuration. This control allows the user to enable or disable that membership. While enabled,
the user can receive hunt group calls when logged in.
Details
• In addition to the lamp indication below, phones display G when any group membership is
enabled.
• Action: Advanced | Hunt Group | Hunt Group Enable.
• Action Data: Group number or name or blank for all groups of which the user is a member.
• Default Label: HGEna or HG Enable.
• Toggles: Yes.
• Status Indication: Required.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Hunt Group Disable

This function is obsolete, the Hunt Group Enable function being able to toggle membership
between enabled and disabled and providing lamp indication of when membership is enabled.
An individual user's membership of any particular hunt groups is programmed through the system
configuration. This control allows the user to disable that membership. They will no longer receive
calls to that hunt group until their membership is enabled again.
Details
• Action: Advanced | Hunt Group | Hunt Group Disable.
• Action Data: Group number or blank for all groups of which the user is a member.
• Default Label: HGDis.

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Button Programming Actions

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Inspect
Supported for CTI emulation only.
Allows users on display phones to determine the identification of held calls. Allows users on an
active call to display the identification of incoming calls.
Details
• Action: Emulation | Inspect.
• Action Data: None.
• Default Label: Inspt.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Internal Auto-Answer
This function is also known as handsfree auto-answer. It sets the user's extension to automatically
connect internal calls after a single tone. This function should only be used on phones that support
handsfree operation.
Details
• Action: Emulation | Internal Auto-Answer.
• Action Data: Optional.
- If left blank this function acts as described above for internal auto-answer.
- FF can be entered. In that case the button will enable/disable headset force feed operation
for external calls. In this mode, when headset mode is selected but the phone is idle, an
incoming external call will cause a single tone and then be automatically connected. This

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 Last Number Redial

operation is only supported on Avaya phones with a fixed HEADSET button. Ring delay is
applied if set on the appearance button receiving the call before the call is auto-connected.
• Default Label: HFAns or Auto Answer.
• Toggles: Yes.
• Status Indication: Required.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Last Number Redial

This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same last number redial process as dialing Feature 5.
Details
• Action: Advanced | Call | Last Number Redial.
• Action Data: None.
• Default Label: Again.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- M-Series and T-Series.
• The button is equivalent to Feature 5.

Leave Word Calling

Supported for CTI emulation only.

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Button Programming Actions

Leaves a message for the user associated with the last number dialed to call the originator.
Details
• Action: Emulation | Leave Word Calling.
• Action Data: None.
• Default Label: LWC.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Line Appearance
Creates an line appearance button linked to the activity of a specified line appearance ID number.
The button can then be used to answer and make calls on that line. For details, see Line
Appearance Buttons on page 1176.
The line appearance button user must also have at least one call appearance button programmed
before line appearance buttons can be programmed.
Line appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Details
• Action: Appearance | Line Appearance.
• Action Data: Line ID number.
• Default Label: Line <Line ID number>.
• Toggles: No.
• Status Indication: Yes.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. Not supported on T7000, T7100, M7100 and M7100N phones.

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 MADN Call Appearance

MADN Call Appearance

Multiple Appearance Directory Number (MADN) emulates an Avaya Communication Server 1000
key and lamp style feature.
When using normal appearance buttons to answer or make calls, the information (name and
number) presented to the other end of the call is that of the button user (subject to any other line
and short code settings). When using a MADN call appearance button, the information presented
is that of the user to which the button is associated rather than that of the button user.
The user associated with a MADN button does not need to have a license or an active extension.
However, they must have an extension number. The system considers the user’s records when
the user makes a call using the MADN buttons. You can have up to 30 MADN buttons associated
with the same user.
MADN can operate in two modes:
• MADN Single Call Appearance (SCA)
The button is configured with the user name of the associated user and one of their call
appearances. This provides the following behaviors:
- Incoming extension calls: The button acts like a Bridged Appearance button to the
associated user.
- Incoming group calls: The button alerts if associated user is a member of the hunt group
and alerting.
- Outgoing calls: The button acts like a Call Appearance. It presents the call as originating
from the button user but with the number and name of the associated user in the calling
party information.
• MADN Multiple Call Appearance (MCA)
The button is configured with just the user name of the associated user. This provides the
following behaviors:
- Incoming extension calls: The button acts like a Coverage Appearance to the associated
user.
- Incoming groups calls: The button does not alert.
- Outgoing calls: The button acts like a Call Appearance. It presents the call as originating
from the button user but with the number of the associated user in the calling party
information.
Details
• Action Either:
- Appearance | MADN Single Call Appearance
- Appearance | MADN Multiple Call Appearance
• Action Data:
- MADN Single Call Appearance: User Name, Call Appearance button number and Ring
Delay.

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Button Programming Actions

- MADN Multiple Call Appearance: User Name and Ring Delay.

• Default Label:
- MADN SCA: <MADN number S=>
- MADN MCA: <MADN number M=>
• Toggles: No.
• Status Indication:
- MADN SCA: Yes. See Bridge Appearance Button Indication.
- MADN MCA: Yes. See Coverage Button Indication.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

Manual Exclude
Supported for CTI emulation only.
Details
• Action: Emulation | Manual Exclude
• Action Data: None.
• Default Label: Excl.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

MCID Activate
This action is used with ISDN Malicious Caller ID call tracing. It is used to trigger a call trace at the
ISDN exchange. The call trace information is then provided to the appropriate legal authorities.
This option requires the line to the ISDN to have MCID enabled at both the ISDN exchange and
on the system. The user must also be configured with Can Trace Calls (User | Telephony |
Supervisor Settings) enabled.

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 Monitor Analog Trunk MWI

Currently, in Server Edition network, MCID is only supported for users using an MCID button and
registered on the same IP500 V2 Expansion system as the MCID trunks.
Details
• Action: Advanced | Miscellaneous | MCID Activate.
• Action Data: None.
• Default Label: MCID or Malicious Call.
• Toggles: No.
• Status Indication: Yes.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Monitor Analog Trunk MWI

Enables a user to receive message waiting indicator (MWI) signals from analog trunks terminating
on the ATM4U-V2 card. MWI is a telephone feature that turns on a visual indicator on a telephone
when there are recorded messages.
Details
• Action: Advanced | Voicemail | Monitor Analog Trunk MWI.
• Action Data: The line appearance ID of the analog line for which MWI will be received.
• Default Label: Trunk MWI.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

Off Hook Station

Enables the user's extension to be controlled by an application, for example SoftConsole. Calls
can then be answered and cleared through the application without having to manually go off or on
hook. Requires the phone to support full handsfree operation.

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Button Programming Actions

Details
• Action: Advanced | Miscellaneous | Off Hook Station.
• Action Data: None.
• Default Label: OHStn.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Pause Recording
This feature can be used to pause any call recording. It can be used during a call that is being
recorded to omit sensitive information such as customer credit card information. This feature can
be used with calls that are recorded both manually or calls that are recorded automatically.
The button status indicates when call recording has been paused. The button can be used to
restart call recording. The system Auto Restart Paused Recording (System | Voicemail) setting
can be used to set a delay after which recording is automatically resumed.
If the voicemail system is configured to provide advice of call recording warnings, then pausing
the recording will trigger a "Recording paused" prompt and a repeat of the advice of call recording
warning when recording is resumed.
Details
• Action: Advanced | Call | Pause Recording.
• Action Data: None.
• Default Label: PauseRec or Pause Recording.
• Toggles: Yes.
• Status Indication: Yes.
• User Admin: No.

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 Priority Call

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

Priority Call
This feature allows the user to call another user even if they are set to 'do not disturb'. A priority
call will follow forward and follow me settings but will not go to voicemail.
Details
• Action: Advanced | Call | Priority Call.
• Action Data: User number or name.
• Default Label: PCall or Priority Call.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- This button action is also supported by the Vantage Connect Expansion application.

Priority Calling
Supported for CTI emulation only.
Details
• Action: Emulation | Priority Calling.
• Action Data: None.
• Default Label: Pcall.
• Toggles: No.
• Status Indication: No.

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Button Programming Actions

• Phone Support: The following table indicates phones which support the programmable
button:
- 1400 Series and 1600 Series.

Private Call
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot
be Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
If enabled during a call, any current recording, intrusion or monitoring is ended.
Details
• Action: Advanced | Call | Private Call.
• Action Data: None.
• Default Label: PrivC or Private Call.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Relay Off
Opens the specified switch in the system's external output port (EXT O/P).
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.

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 Relay On

Details
• Action: Advanced | Relay | Relay Off.
• Action Data: Switch number (1 or 2).
• Default Label: Rely-.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Relay On
Closes the specified switch in the system's external output port (EXT O/P).
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Details
• Action: Advanced | Relay | Relay On.
• Action Data: Switch number (1 or 2).
• Default Label: Rely+ or Relay On.
• Toggles: Yes.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Relay Pulse
Closes the specified switch in the system's external output port (EXT O/P) for 5 seconds and then
opens the switch.

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Button Programming Actions

This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Details
• Action: Advanced | Relay | Relay Pulse.
• Action Data: Switch number (1 or 2).
• Default Label: Relay or Relay Pulse.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Resume Call
Resume a call previously suspended to the specified ISDN exchange slot. The suspended call
may be resumed from another phone/ISDN Control Unit on the same line.
Details
• Action: Advanced | Call | Resume Call.
• Action Data: ISDN Exchange suspend slot number.
• Default Label: Resum.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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 Request Coaching Intrusion

Request Coaching Intrusion

This feature allows a user to request that another user intrude on a call and talk to them without
being heard by the other call parties to which they can still talk.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
• Intrusion features uses system conference resources during the call. If insufficient conference
resource are available, the feature cannot be used.
Warning:
• Listening to a call without the other parties being aware is subject to local regulations.
You must ensure that you have complied with the local regulations. Failure to do so can
result in penalties.
•
The system support a range of other call intrusion methods in addition to this feature. The Request
Coaching Intrusion feature exhibits the following behavior:
• A coaching request can be sent to a user or a group.
• While the request is pending, the user can cancel the request by pressing the Request
Coach button again.
• Once a coaching session is established, the user that initiated the request can include the
coach in the call, transfer the call to the coach, or drop the coach from the call.
• Once a coaching session is established, the coach can join the call or steal the call. The
coach cannot transfer or conference the call.
• Once the primary call ends, the coaching call continues.
Details
Details
• Action: Advanced | Call | Request Coaching Intrusion.
• Action Data: None.
• Default Label: Request Coach or Request Coaching Intrusion.
• Toggles: Yes.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

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Button Programming Actions

Retrieve Call
Retrieves a call previously held to a specific ISDN exchange slot. Only available when supported
by the ISDN exchange.
Details
• Action: Advanced | Call | Retrieve Call.
• Action Data: Exchange hold slot number.
• Default Label: Retriv.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Ring Back When Free

Sets a ringback on the extension being called. When the target extension ends its current call, the
ringback users is rung (for their set No Answer Time) and if they answer, a new call is made to the
target extension.
Ringback can be cleared using the Cancel Ring Back When Free function.
Details
• Action: Advanced | Miscellaneous | Ring Back When Free.
• Action Data: None.
• Default Label: AutCB or Auto Callback.
• Toggles: No.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

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 Ringer Off

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. The button is equivalent to Feature 2.
- This button action is also supported by the Vantage Connect Expansion application.

Ringer Off
Switches the phone's call alerting ring on/off.
Details
• Action: Emulation | Ringer Off.
• Action Data: None.
• Default Label: RngOf or Ringer Off.
• Toggles: Yes.
• Status Indication: Yes Required.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Self-Administer
Allows a user to program features against other programmable buttons themselves.
Appearance can no longer be used to create call appearance buttons. Similarly, existing call
appearance button cannot be overwritten using any of the other Admin button functions.
User's with a log in code will be prompted to enter that code when they use this button action.

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Button Programming Actions

On 4412D+, 4424D+, 6408D, 6416D, 6424D phones:

• Admin can be permanently accessed via Menu , , , Admin. See Using a Menu Key.
• Admin1 can be permanently accessed via Menu , Menu , , ProgA, , , DSS.
Details
• Action: Emulation | Self-Administer.
• Action Data: See below.
Value T-Series and M-Series phones Other Phones
None The Feature *3 process is If no value is set, the button allows user
started with an alternate set of programming of the following emulation actions:
possible functions.
- Abbreviated Dial, Abbreviated Dial Program,
Account Code Entry, AD Suppress,
Automatic Callback, Break Out, Call
Forwarding All, Call Park, Call Park and
Page, Call Park To Other Extension, Call
Pickup, Call Pickup Any, Conference Meet
Me, Dial Paging, Directed Call Pickup,
Directory, Drop, Group Paging, Headset
Toggle, Hook Flash, Internal Auto-Answer,
Ringer Off, Self-Administer, Send All Calls,
Set Absent Text, Set Hunt Group Night
Service, Time of Day, Timer, Twinning.
1 The Feature *1 process If 1 is entered as the telephone number,
is started for assigning allows user programming of the following system
Abbreviated Dial button. functions.
- Abbreviated Dial, Group, CPark, User, Flash
Hook.
2 The Feature *6 process is If 2 is entered, the button can be used for viewing
started for setting the ring type. details of the control unit type and its software
version. This option is available. If the user has a
log in code set, they will be prompted to enter
that code. System phone users (see System
Phone Features on page 728) can also use the
button to manually set the system's date and
time.
3 The option 3 is used with M- Not used.
Series and T-Series sets to
enable display contrast control.

• Default Label: Admin or Self Administer.

• Toggles: No.
• Status Indication: No.
• User Admin: Yes.

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 Send All Calls

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Send All Calls

Sets the user's extension into 'Do Not Disturb' mode. Callers, other than those on the user's do not
disturb exception list, receive busy or are diverted to the users voicemail mailbox. Note that with
a call already connected and other calls already alerting, enabling Do Not Disturb will not affect
those calls already existing. For full details of see Do Not Disturb.
When on, most phones display an N on the display. This function and the Do Not Disturb On
function work in parallel, ie. setting one sets the other.
Details
• Action: Emulation | Send All Call.
• Action Data: None.
• Default Label: SAC or Send All Calls.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Absent Text

This feature can be used to select the user's current absence text. This text is then displayed
to internal callers who have suitable display phones or applications. It doesn't changes the users

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Button Programming Actions

status. The absence text message is limited to 128 characters. Note however that the amount
displayed will depend on the caller's device or application.
The text is displayed to callers even if the user has forwarded their calls or is using follow me.
Absence text is supported across a multi-site network.
The user still has to select Set or Clear on their phone to display or hide the text.
Details
• Action: Advanced | Set | Set Absent Text.
• Action Data: Optional. On certain phones, if the button is set without any Action Data, the
user is prompted to select their absence text and switch it on/off through a menu shown on
the phone display.
The telephone number should take the format "y,n,text" where:
- y = 0 or 1 to turn this feature off or on respectively.
- n = the number of the absent statement to use:
0 = None. 4 = Meeting until. 8 = With cust. til.
1 = On vacation until. 5 = Please call. 9 = Back soon.
2 = Will be back. 6 = Don't disturb until. 10 = Back tomorrow.
3 = At lunch until. 7 = With visitors until. 11 = Custom.

text = any text to follow the absent statement..

• Default Label: Absnt or Absence Text.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Account Code

Dials an account code and then returns dial tone for the user to dial a number. Can also be used
to enter an account code after a call has been connected.
Details
• Action: Advanced | Set | Set Account Code..
• Action Data: Account code or blank. If blank, the user is prompted to dial an account code
after pressing the button. This option is not supported on XX02 phone modules.

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 Set Hunt Group Night Service

• Default Label: Acct or Account Code.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Hunt Group Night Service

Puts the specified hunt group into Night Service mode. Calls to a group set to night service,
receive busy or are diverted to voicemail if available or are diverted to the group's night service
fallback group if set.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Details
• Action: Advanced | Set | Set Hunt Group Night Service.
• Action Data: Hunt group extension number.
- If left blank, the button will affect all hunt groups of which the user is a member.
- The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively. The
service is indicated by setting the service name as the telephone number or action data.
Do not use quotation marks.
• Default Label: HGNS+ or HG Night Service.
• Toggles: Yes.
• Status Indication: Required. If the button is blank (no specific hunt group) it will indicate on
if any one of the hunt groups of which the user is a member is set to night service. If the
button is set for multiple hunt groups it will indicate on if any one of those groups is set to
night service.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

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Button Programming Actions

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Hunt Group Out Of Service

Puts the specified hunt group into Out of Service mode. Calls to a group set to out of service
receive busy or are diverted to voicemail if available or are diverted to the group's out of service
fallback group if set.
This function can be used to used to override hunt groups already set to night service mode by an
associated time profile.
Details
• Action: Advanced | Set | Set Hunt Group Out of Service.
• Action Data: Hunt group extension number. If left blank, the button will affect all hunt groups
of which the user is a member.
• Default Label: HGOS+ or HG Out of Service.
• Toggles: Yes.
• Status Indication: Required. If the button is blank (no specific hunt group) it will indicate on if
any one of the hunt groups of which the user is a member is set out of service. If the button is
set for multiple hunt groups it will indicate on if any one of those groups is set out of service.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

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 Set Inside Call Seq

Set Inside Call Seq

This feature allows the user to select the ringing used on their analog extension for internal calls.
Details
• Action: Advanced | Set | Set Inside Call Seq.
• Action Data: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.
• Default Label: ICSeq.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

Set Night Service Destination

This button allows the user to change the Night Service target of a hunt group. The button user
does not have to be a member of the hunt group. In a multi-site network this function can be used
for hunt groups on remote systems.
Changing the destination does not affect calls already ringing at the hunt groups previous night
service destination.
Details
• Action: Advanced | Set | Set Night Service Group.
• Action Data: Hunt group extension number. This is the group for which the night service
destination is being set.
• Default Label: SetNSG or HG NS Group.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

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Button Programming Actions

Set No Answer Time

Allows the user to change their no answer time setting. This is the time calls ring before going to
voicemail or following the user's divert on no answer setting if set on.
In situations where call coverage is also being used, the user's no answer time must be greater
than their individual coverage time for coverage to occur.
Details
• Action: Advanced | Set | Set No Answer Time.
• Action Data: Time in seconds.
• Default Label: NATim or No Answer Time.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Out of Service Destination

This button allows the user to change the Out of Service target of a hunt group. The button user
does not have to be a member of the hunt group. In a multi-site network this function can be used
for hunt groups on remote systems.
Changing the destination does not affect calls already ringing at the hunt groups previous Out of
Service destination.
Details
• Action: Advanced | Set | Set Out of Service Group.
• Action Data: Hunt group extension number. This is the group for which the night service
destination is being set.
• Default Label: SetOOSG or HG OS Group.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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 Set Outside Call Seq

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Set Outside Call Seq

This feature allows the user to select the ringing used on their analog extension for external calls.
Details
• Action: Advanced | Set | Set Outside Call Seq.
• Action Data: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.
• Default Label: OCSeq.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

Set Ringback Seq

This feature allows the user to select the ringing used on their analog extension for ringback calls.
Details
• Action: Advanced | Set | Set Ringback Seq.
• Action Data: 0 to 10.
- The number sets to the required ring pattern. See Ring Tones on page 655.
- The numbering starts at 0 for Default Ring, 1 for Ring Normal, 2 for RingType1, and so on.
• Default Label: RBSeq.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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Button Programming Actions

Set Wrap Up Time

Allows users to change their Wrap-up Time (User | Telephony | Call Settings) setting. Other
phones or applications monitoring the user's status will indicate the user as still being busy (on a
call). Hunt group calls will not be presented to the user.
If the user is using a single line set, direct calls also receive busy treatment. If the user is using a
mutli-line set (multiple call appearances), direct calls to them will ring as normal.
It is recommended that this option is not set to less than the default of 2 seconds. 0 is used to
allow immediate ringing.
Details
• Action: Advanced | Set | Set Wrap Up Time.
• Action Data: Time in seconds. Range 0 to 99999 seconds.
• Default Label: WUTim or Wrap-up Time.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Speed Dial
When pressed, the button invokes the same process as dialing Feature 0.
• If Feature 0 is followed by a 3-digit index number in the range 000 to 999, the system
directory entry with the matching index number is dialed.
• If Feature 0 is followed by * and a 2-digit index number in the range 00 to 99, the personal
directory entry with the matching index number is dialed. Note: Release 10.0 allows users to
have up to 250 personal directory entries. However, only 100 of those can be assigned index
numbers.
Details
• Action: Advanced | Dial | Speed Dial.
• Action Data: None.
• Default Label: SpdDial.
• Toggles: No.

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 Stamp Log

• Status Indication: No.

• User Admin: No.
• Phone Support
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Stamp Log
The stamp log function is used to insert a line into any System Monitor trace that is running. The
line in the trace indicates the date, time, user name and extension plus additional information. The
line is prefixed with LSTMP: Log Stamped and a log stamp number. When invoked from a Avaya
phone with a display, Log Stamped# is also briefly displayed on the phone. This allows users to
indicate when they have experienced a particular problem that the system maintainer want them
to report and allows the maintainer to more easily locate the relevant section in the monitor trace.
The log stamp number is set to 000 when the system is restarted. The number is then
incremented after each time the function is used in a cycle between 000 and 999. Alternately
if required, a specific stamp number can be assigned to the button or short code being used for
the feature.
Details
• Action: Advanced | Miscellaneous | Stamp Log.
• Action Data: Optional. Blank or any 3 digit number.
• Default Label: Stamp Log.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. Not supported on T7000, T7100, M7100 and M7100N telephones.
- 1100 Series and 1200 Series.

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Button Programming Actions

Stored Number View

Supported for CTI emulation only.
Allows a user to view the contents of any programmed feature button.
Details
• Action: Emulation | Stored Number View.
• Action Data: None.
• Default Label: BtnVu.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Suspend Call
Uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The call is placed in exchange slot 0 if a slot number is not specified. Only
available when supported by the ISDN exchange.
Details
• Action: Advanced | Suspend | Suspend.
• Action Data: Exchange slot number or blank (slot 0).
• Default Label: Suspe.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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 Suspend CW

Suspend CW
Uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange and answer
the call waiting. The call is placed in exchange slot 0 if a slot number is not specified. Only
available when supported by the ISDN exchange.
Details
• Action: Advanced | Suspend | Suspend CW.
• Action Data: Exchange slot number or blank (slot 0).
• Default Label: SusCW.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Swap CLID Name/Number

Allows the user to toggle between Caller Name and Caller ID.
Details
• Action: Emulation | Swap CLID Name/Number
• Action Data: None.
• Default Label:
• Toggles: Yes.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- M-Series and T-Series.

Time of Day
Displays the time and date on the user's telephone. This function is ignored on those Avaya
phones that display the date/time by default.

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Button Programming Actions

Details
• Action: Emulation | Time of Day.
• Action Data: None.
• Default Label: TmDay.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: Yes.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Time Profile
You can manually override a time profile. The override settings allow you to mix timed and manual
settings.
The button indicator will show the Time Profile state and pressing the button will present a menu
with five options and an indication of the current state. The menu options are listed below.
Menu Option Description
Timed Operation No override. The time profile operates as configured.
Active Until Next Timed Use for time profiles with multiple intervals. Select to make the current timed
Inactive interval active until the next inactive interval.
Inactive Until Next Use for time profiles with multiple intervals. Select to make the current active timed
Timed Active interval inactive until the next active interval.
Latch Active Set the time profile to active. Timed inactive periods are overridden and remain
active.
Latch Inactive Set the time profile to inactive. Timed active periods are overridden and remain
inactive.

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 Time Profile

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

Details
• Action: Emulation | Time Profile
• Action Data: Time profile name.
• Default Label: TP or Time Profile
• Toggles: No.
• Status Indication:
Status 1400, 1600, 9608, 9611, J100 9621, 9641
On Green Green On Green
Off Off Off Grey

• User Admin: No
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.

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Button Programming Actions

Timer
Starts a timer running on the display of the user's extension. The timer disappears when the user
ends a call.
This function can be used on Avaya phones (except 9600 Series) that display a call timer next
to each call appearance. The button will temporarily turn the call timer on or off for the currently
selected call appearance. The change only applies for the duration of the current call.
• Action: Emulation | Timer.
• Action Data: None.
• Default Label: Timer.
• Toggles: Yes.
• Status Indication: No.
Details
• User Admin: Yes.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Transfer
This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same transfer process as dialing Feature 70.
Details
• Action: Advanced | Call | Transfer.
• Action Data: None.
• Default Label: Xfer.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.

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 Toggle Calls

Toggle Calls
Cycle between the user's current call and any held calls.
Details
• Action: Advanced | Call | Toggle Calls..
• Action Data: None.
• Default Label: Toggl.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

Twinning
This action can be used by user's setup for mobile twinning. This action is not used for internal
twinning.
While the phone is idle, the button allows the user to set and change the destination for their
twinned calls. It can also be used to switch mobile twinning on/off and indicates the status of that
setting.
When a call has been routed by the system to the user's twinned destination, the Twinning button
can be used to retrieve the call at the user's primary extension.
In configurations where the call arrives over an IP trunk and the outbound call is on an IP trunk,
multi-site network may optimise the routing and in this case the button may not be usable to
retrieve the call.
Mobile Twinning Handover When on a call on the primary extension, pressing the Twinning
button will make an unassisted transfer to the twinning destination. This feature can be used even
if the user's Mobile Twinning setting was not enabled.
During the transfer process the button will wink. Pressing the twinning button again will halt the
transfer attempt and reconnect the call at the primary extension.
The transfer may return if it cannot connect to the twinning destination or is unanswered within the
user's configured Transfer Return Time (if the user has no Transfer Return Time configured, a
enforced time of 15 seconds is used).
Details
• Action: Emulation | Twinning.

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Button Programming Actions

• Action Data: None.

• Default Label: Twinning.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 M-Series, T-
Series
- On. Green on Green on Green On
- Off. Off Off Grey Off
- Twinned call at Red on Red on Blue On
secondary

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.
- This button action is also supported by the Vantage Connect Expansion application.

Unpark Call
This function is obsolete, since the Call Park function can be used to both park and retrieve calls
and provides visual indication of when calls are parked. Retrieve a parked call from a specified
system park slot.
Details
• Action: Advanced | Call | Unpark Call.
• Action Data: System park slot number. This must match a park slot ID used to park the call.
• Default Label: UnPark.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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 User

User
Monitors whether another user's phone is idle or in use. The Telephone Number field should
contain the users name enclosed in double quotes. The button can be used to make calls to the
user or pickup their longest waiting call when ringing. On buttons with a text label, the user name
is shown.
The actions performed when the button is pressed will depend on the state of the target user and
the type of phone being used. It also depend on whether the user is local or on a remote multi-site
network system.
Phone Large display 1400, 1600, 9500, 9600, M-Series Other Phones or across a
and T-Series Phones multi-site network
Idle Call the user. Whilst ringing the phone displays options to Callback (set an
automatic callback ) and Drop (end the call attempt).
Ringing • Call Pickup: Pickup the ringing call. Picks up the call.
• Call: Make a call to the user.
On a Call The following options are displayed (name lengths Call, Voicemail and
may vary depending on the phone display): Callback options are
supported.
• Call: Make a call to the user.
• Message: Cause a single burst of ringing on
the target phone. On some phones, when they
end their current call their phone will then display
PLEASE CALL and your extension number.
• Voicemail: Call the user's voicemail mailbox.
• Callback: Set an automatic callback.
• Drop Disconnect the user's current call.
• Acquire: Shown if able to intrude on the user. Take
control of the call.
• Intrude: Shown if able to intrude on the user.
Intrude into the call, turning it into a 3-way
conference.
• Listen: Shown if configured to be able to listen
to (monitor) the user. Start silent monitoring of the
user's call.

A User button can be used in conjunction with other buttons to indicate the target user when those
buttons have been configured with no pre-set user target. In cases where the other button uses
the phone display for target selection this is only possible using User buttons on an associate
button module.
The following changes have been made to the indication of user status via BLF (busy lamp field)
indicators such as a User button:

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Button Programming Actions

The status shown for a logged out user without mobile twinning will depend on whether they have
Forward Unconditional enabled.
• If they have Forward Unconditional enabled the user is shown as idle.
• If they do not have Forward Unconditional enabled they will show as if on DND.
The status shown for a logged out user with mobile twinning will be as follows:
• If there are any calls alerting or in progress through the system to the twinned destination, the
user status is shown as alerting or in-use as appropriate. This includes the user showing as
busy/in-use if they have such a call on hold and they have Busy on Held enabled.
• If the user enables DND through Mobile Call Control or one-X Mobile client, their status will
show as DND.
• Calls from the system direct to the user's twinned destination number rather than redirected
by twinning will not change the user's status.
Details
• Action: User.
• Action Data: User name enclosed in "double-quotes".
• Default Label: <the user name>.
• Toggles: No.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series, M-
Series
- Idle. Off Off Grey Off
- Alerting. Red flash Red flash Blue Slow flash
- In Use/Busy. Red wink Red wink Blue Fast flash
- DND Red on Red on Green On

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Visual Voice
This action provides the user with a menu for access to voicemail mailboxes. The menu provides
the user with options for listening to messages, leaving messages and managing the mailbox. If

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 Visual Voice

no action data is specified, then it is the user's mailbox. Action Data can be used to specify the
mailbox of another user or group.
Note:
You can also use the “H” and “U” user source numbers to add another mailbox to your Visual
Voice menu. See User | Source Numbers
If the Action Data has been configured, pressing the button for an incoming call or while a call
is connected sends the call to the user mailbox specified in the action data. If no Action Data is
configured, the user is prompted to enter a mailbox.
On phones that have a display but do not support full visual voice operation as indicated below,
use the button for user mailbox access using voice prompts and for direct to voicemail transfer
during a call is supported.
Access to Visual Voice on supported phones can be triggered by the phone's MESSAGES button
rather than requiring a separate Visual Voice programmable button. This is done using the option
System | Voicemail | Messages button goes to Visual Voice.
Details
• Action: Emulation | Visual Voice.
• Action Data: All local users and groups and all users and groups on systems in the network,
except for the user on which the button is being programmed.
• Default Label: Voice.
• Toggles: No.
• Status Indication: When action data is configured, the status lamp provides a message
waiting indicator for the monitored mailbox.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
1. Takes the user direct to the listen part of Visual Voice. For the full Visual Voice menu
options, the user should use Menu | Settings | Voicemail Settings.
Visual Voice Controls
The arrangement of options on the screen will vary depending on the phone type and display size.
Option Description
Listen Access your own voicemail mailbox. When pressed the screen will show the number
of New, Old and Saved messages. Select one of those options to start playback of
messages in that category. Use the up arrow and arrow keys to move through the
message. Use the options below.
Listen Play the message.
Table continues…

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Button Programming Actions

Option Description
Pause Pause the message playback.
Delete Delete the message.
Save Mark the message as a saved message.
Call Call the message sender if a caller ID is available.
Copy Copy the message to another mailbox. When pressed a number of additional options
are displayed.
Message Record and send a voicemail message to another mailbox or mailboxes.
Greeting Change the main greeting used for callers to your mailbox. If no greeting has been
recorded then the default system mailbox greeting is used.
Mailbox Name Record a mailbox name. This feature is only available on systems using Embedded
Voicemail.
Email This option is only shown if you have been configured with an email address for
voicemail email usage in the system configuration. This control allows you to see and
change the current voicemail email mode being used for new messages received by
your voicemail mailbox. Use Change to change the selected mode. Press Donewhen
the required mode is displayed. Possible modes are:
Password Change the voicemail mailbox password. To do this requires entry of the existing
password.
Voicemail Switch voicemail coverage on/off.

Voicemail Collect
Connects to the voicemail server. The telephone number must indicate the name of the Voicemail
box to be accessed, eg. "?Extn201" or "#Extn201". The ? indicates "collect Voicemail" and the
# indicates "deposit Voicemail". This action is not supported by voicemail using Intuity emulation
mode.
When used with Voicemail Pro, names of specific call flow start points can also be used to directly
access those start points via a short code. In these cases ? is not used and # is only used if
ringing is required before the start points call flow begins.
Details
• Action: Advanced | Voicemail | Voicemail Collect.
• Action Data: See above.
• Default Label: VMCol or VMail Collect.
• Toggles: No.
• Status Indication: No.
• User Admin: No.

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 Voicemail Off

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
1. For access to the users own mailbox, this button is equivalent to Feature 65 and
Feature 981.

Voicemail Off
Disables the user's voicemail box from answering calls that ring unanswered at the users
extension. This does not disable the user's mailbox and other methods of placing messages into
their mailbox.
This button function is obsolete as the Voicemail On function toggles on/off.
Details
• Action: Advanced | Voicemail | Voicemail Off.
• Action Data: None.
• Default Label: VMOff.
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.

Voicemail On
Enables the user's voicemail mailbox to answer calls which ring unanswered or arrive when the
user is busy.
Details
• Action: Advanced | Voicemail | Voicemail On.

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Button Programming Actions

• Action Data: None.

• Default Label: VMOn or VMail On.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.
- 1100 Series and 1200 Series.
- This button action is also supported by the Vantage Connect Expansion application.

Voicemail Ringback Off

Disables voicemail ringback to the user's extension. This button function is obsolete as the
Voicemail Ringback On function toggles on/off.
Details
• Action: Advanced | Voicemail | Voicemail Ringback Off.
• Action Data: None.
• Default Label: VMRB-
• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 1400 Series and 1600 Series.

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 Voicemail Ringback On

Voicemail Ringback On
Enables voicemail ringback to the user's extension. Voicemail ringback is used to call the user
when they have new voicemail messages in their own mailbox or a hunt group mailbox for which
they have been configured with message waiting indication.
The ringback takes place when the user's phone returns to idle after any call is ended.
Details
• Action: Advanced | Voicemail | Voicemail Ringback On.
• Action Data: None.
• Default Label: VMRB+ or VMail Ringback.
• Toggles: Yes.
• Status Indication: Yes.
Status 1400, 1600, 9500 9608, 9611, J100 9621, 9641 T-Series,
On Green on Green on Green On
Off Off Off Grey Off

• User Admin: No.

• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
- M-Series and T-Series.

Whisper Page
This feature allows you to intrude on another user and be heard by them without being able to
hear the user's existing call which is not interrupted.
For example: User A is on a call with user B. When user C intrudes on user A, they can be heard
by user A but not by user B who can still hear user A. Whisper page can be used to talk to a user
who has enabled private call.
• Intrusion features are controlled by the Can Intrude setting of the user intruding and the
Cannot Be Intruded setting of user being intruded on. By default, no users can intrude and
all users cannot be intruded.
The system support a range of other call intrusion methods in addition to this feature.
Details
• Action: Advanced | Call | Whisper Page.
• Action Data: User number or name or blank for entry when pressed.

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Button Programming Actions

• Default Label: Whisp or Whisper Page.

• Toggles: No.
• Status Indication: No.
• User Admin: No.
• Phone Support: Note that support for particular phone models is also dependent on the
system software level.
- 9500 Series, 9600 Series and J100 Series.
- 1400 Series and 1600 Series.
1. Not supported on non-IP telephones when using a headset.

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Part 13: Call Appearance Buttons

Appearance Buttons

Many Avaya phones supported on system have a programmable keys or buttons (the terms 'key'
and 'button' mean the same thing in this context). A wide range of actions can be assigned to those
buttons, see Button Programming Actions on page 1046.
These actions can be assigned to the programmable buttons on a user's phone. Those 'appearance'
buttons can then be used to answer, share, switch between and in some case make calls. This type
of call handling is often called 'key and lamp mode'.
The following sections in this documentation relate to a set of button actions collectively called
'appearance' actions. These are:
Appearance Button Description
Type
Call Appearances Call appearance buttons are used to display alerts for incoming calls directed to
a user's extension number or to a hunt group of which they are a member. Call
appearance buttons are also used to make outgoing calls.
By having several call appearance buttons, a user is able to be alerted about
several calls, select which call to answer, switch between calls and take other
actions.
See Call Appearance Buttons on page 1161.
Bridged Appearances A bridged appearance button shows the state of one of another user's call
appearance buttons. It can be used to answer or join calls on that user's call
appearance button. It can also be used to make a call that the call appearance
user can then join or retrieve from hold.
See Bridged Appearance Buttons on page 1166.
Line Appearances Call coverage allows a user to be alerted when another user has an unanswered
call.
See Line Appearance Buttons on page 1176.
Table continues…

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Call Appearance Buttons

Appearance Button Description

Type
Call Coverage Line appearance buttons allow specific individual line to be used when making
Appearances calls or answered when they have an incoming call. It also allows users to bridge
into calls on a particular line.
See Call Coverage Buttons on page 1171.

Note:
• For all the examples within this documentation, it is assumed that Auto Hold is on and
Answer Pre-Select is off unless otherwise stated.
• The text shown on phone displays in the examples are typical and may vary between
phone types, locales and system software releases.

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Chapter 104: Call Appearance Buttons

Call appearance buttons are used to display alerts for incoming calls directed to a user's extension
number or to a hunt group of which they are a member. Call appearance buttons are also used to
make outgoing calls.
By having several call appearance buttons, a user is able to be alerted about several calls, select
which call to answer, switch between calls and take other actions.

When all the user's call appearance buttons are in use or alerting, any further calls to their extension
number receive busy treatment. Instead of busy tone, the user's forward on busy is used if enabled
or otherwise voicemail if available.
Call appearance buttons are the primary feature of key and lamp operation. None of the
other appearance button features can be used until a user has some call appearance button
programmed[1].
There are also addition requirements to programming call appearance buttons:
• Call appearance buttons must be the first button programmed for the user.
• Programming a single call appearance button for a user is not supported. The normal default is
3 call appearances per user except on phones where only two physical buttons are available.
Related links
Call Appearance Example 1 on page 1162
Call Appearance Example 2 on page 1162
How are Call Appearance Buttons Treated? on page 1163
Call Appearance Button Indication on page 1164

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Call Appearance Buttons

Call Appearance Example 1

In this example, the user has multiple call appearance buttons.
Phone Idle The phone is currently idle.

First Call Alerts

A call arrives. It alerts against the first available call appearance button. Pressing
that button will answer the call.

Call Answered
The call is now connected.

Second Call Alerts

A second call arrives whilst the first is still connected. It alerts against the next
available call appearance button. As the user has a call in progress, the alert gives
just a single ring and briefly display details of the caller.
Pressing the Second Call Appearance
Pressing the second call appearance button will hold the first call and answer the
second.

Related links
Call Appearance Buttons on page 1161

Call Appearance Example 2

In this example, the user will use their call appearances to make two calls and start a conference
between those calls.
Initial Call
The user has a call in progress, shown on their first call appearance button. It is
decided to conference another user into the call.

Make Conference Enquiry

Pressing the CONFERENCE button on the users phone automatically places the
current call on hold and takes the phone off hook on the next available call
appearance.
Table continues…

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 How are Call Appearance Buttons Treated?

Enquiry in Progress
The other extension has been dialed and invited to join a conference call. The user
presses the CONFERENCE button on their phone again.

Conference Starts
The conference call has started. The separate call appearances have collapsed to
a single appearance that represents the conference.

Related links
Call Appearance Buttons on page 1161

How are Call Appearance Buttons Treated?

For incoming calls
• Call Waiting settings are ignored except for hunt group call waiting where the call waiting
tone is replaced by an alert on a call appearance button if available.
• Follow Me, Forward Unconditional and Forward Hunt Group Calls are used when set.
• If Do Not Disturb is set, only calls from numbers in the user's Do Not Disturb Exception list
will alert if a call appearance is available.
Busy status
In both cases below, even when busy, the user may still receive alerts on other appearance
buttons.
• For calls direct to the user's extension number The user is busy when all their available
call appearances are in use. Instead of busy tone, the user's forward on busy is used if
enabled or otherwise voicemail if available.
• For calls to a hunt group of which the user is a member The user is busy to further hunt
group calls when they have any appearance button in use on their phone. The only exception
is calls to a collective hunt group with call waiting.
For outgoing calls
• Outgoing calls are treated exactly the same as calls made by non-appearance button users.
• External Calls made on a call appearance, which route out on a line for which the user also
has a line appearance, will remain on the call appearance. The line appearance will indicate
'in use elsewhere'.
For call appearance buttons matched by a bridged appearance button
• If the bridged appearance is used to make or answer calls, the state of the call appearance
will match that of the bridged appearance.
• If the call is put on hold by the bridged appearance user, the call appearance will show 'on
hold elsewhere'.

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Call Appearance Buttons

Other
• Held/Parked Call Timeout If the user has parked a call, the parked call timer only starts
running when the user is idle rather than on another call.
• Incoming calls routed directly to the user as the incoming call routes destination on a line for
which the user also has a line appearance, will only alert on the line appearance. These calls
do not follow any forwarding set but can be covered.
Related links
Call Appearance Buttons on page 1161

Call Appearance Button Indication

On phones with a text display area next to the button, by default a=, b= and so on is displayed.
This can be replaced by another label if required.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of call appearance buttons (alerting, held,
etc) are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay on page 1188.
Icon Button Dual LED Appearance Button State
Button
Red off, Idle
Green off. The call appearance is not in use and is not currently selected.
Red on, Idle + Selected
Green off. The call appearance is not in use but is the current selected button that
will be used if the user goes off hook.
Red off, Alerting
Flashing icon. Green steady The matching call appearance is alerting for an incoming call. This is
flash. accompanied by ringing. If the user is already on a call, only a single ring
is given.
Red on, Alerting + Selected
Flashing icon. Green steady As above but Ringing Line Preference has made this the user's current
flash. selected button.
Red on, In Use Here
Green on. The user has a call connected on the call appearance or is dialing.
Red off, In Use Elsewhere
Green on. The call appearance button is in use on a bridged appearance.
Table continues…

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 Call Appearance Button Indication

Icon Button Dual LED Appearance Button State

Button
Red off, On Hold Here
Green fast flash. The call has been put on hold by this user.
Red fast flash, On Hold Pending Transfer
Green fast flash Applies to 1400, 1600, 9500 and 9600 Series phones.
Red off, On Hold Elsewhere
Green A call on a bridged appearance button matched to the call appearance
intermittent has been put on hold. Calls on a call appearance that are put on hold
flash. by another user will continue to show connected lamp status, though the
phone display will indicate a held call.
Red off, Inaccessible
Icon Green broken The button pressed is not accessible. The call is still dialing, ringing or
flash. cannot be bridged into.
flashes off.

Related links
Call Appearance Buttons on page 1161

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Chapter 105: Bridged Appearance Buttons

A bridged appearance button shows the state of one of another user's call appearance buttons. It
can be used to answer or join calls on that user's call appearance button. It can also be used to
make a call that the call appearance user can then join or retrieve from hold.

When the user's call appearance button alerts, any associated bridged appearance buttons on other
user's phones also alert. The bridged appearance buttons can be used to answer the call on the call
appearance button user's behalf.
When the call appearance button user answers or makes a call, any associated bridged appearance
buttons on other users' phones show the status of the call, ie. active, on hold, etc. The bridged
appearance button can be used to retrieve the call if on hold or to join the call if active (subject to
intrusion permissions).
Note Bridged appearance buttons are different from the action of bridging into a call (joining a call).
See Joining Other Calls (Bridging).
Bridged appearance buttons are not supported between users on different systems in a multi-site
network.
Related links
Bridged Appearance Example 1 on page 1167
Bridged Appearance Example 2 on page 1167
Bridged Appearance Example 3 on page 1168
How are Bridged Appearances Treated? on page 1169
Bridged Appearance Button Indication on page 1170

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 Bridged Appearance Example 1

Bridged Appearance Example 1

In this example, one user is able to see the status of the other user's call appearances, and when
necessary answer calls for the other user. Both users have Ringing Line Preference and Auto
Hold on.
Call Appearance User Bridged Appearance User Both Phone Idle Our user
has bridged appearance buttons
that match a colleague's call
appearances buttons.

First Call The colleague has a

call alerting on their first call
appearance button. It also alerts on
our user's first bridged appearance
button.

Call Answered The colleague

has answered the call. The
bridged appearance indicates 'in
use elsewhere'.

Second Call Another call alerts at

the colleagues phone and again
is mirrored on our user's second
bridged call appearance button.

Call Answered Our user has

gone off hook and answered
the incoming call alerting on the
bridged call appearance.

Related links
Bridged Appearance Buttons on page 1166

Bridged Appearance Example 2

In this example, the bridged appearance user makes a call on behalf of the call appearance user.
Once the call is connected, they put it on hold. The call appearance user is able to take the call
off hold using their call appearance button. Both users have Ringing Line Preference and Auto
Hold on.

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Bridged Appearance Buttons

Call Appearance User Bridge Appearance User Both Phones Idle Our user
has bridged appearance buttons
that match a colleague's call
appearances buttons.

Bridged User Makes Call Our user

has pressed a bridged appearance
and made a call on it. The matching
call appearance shows 'in use
elsewhere'.

Call Put on Hold Having made the

call, the bridged user puts it on
hold. The matching call appearance
indicates 'on hold elsewhere'.

Call Taken Off Hold By pressing

the call appearance, the first user
has answered the held call. The
bridged appearance user returns to
idle.

Related links
Bridged Appearance Buttons on page 1166

Bridged Appearance Example 3

In this example, a call is passed from the call appearance user to the bridged appearance user.
Both users have Ringing Line Preference and Auto Hold on.
Bridged Appearance Call on Colleague's Phone
User
The call appearance user has answered a call on one of their call appearances.
The bridged appearance user's matching bridged appearance shows 'in use
elsewhere'.

Call Held by Colleague

The call appearance user has put the call on hold and called the bridged
appearance user. The first bridged call appearance shows a call 'on hold
elsewhere' whilst the second matches the call between users.
Table continues…

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 How are Bridged Appearances Treated?

Enquiry Call Between Colleagues

By going off hook, the bridged appearance user has answered the call from the
call appearance user. They are asked to pickup the call on the colleagues first call
appearance.
Call Taken Off Hold
Pressing the first bridged appearance button takes that call off hold and connects it
to the bridged appearance user.
In this example, Auto Hold is not set for the system, so pressing the bridged
appearance button disconnected the call from the colleague.
If Auto Hold had been set, the colleague's call would have been put on hold until
they hung up.

Related links
Bridged Appearance Buttons on page 1166

How are Bridged Appearances Treated?

Bridged appearance buttons operate in parallel with their matching call appearance button.
• Whose user settings control the call? Until answered on a bridged appearance button,
calls alerting on a bridged appearance button follow the settings of the user or hunt group to
which the call was originally directed.
• If the call appearance is in use, any matching bridged appearance will indicate the same.
• If a bridged appearance is in use, the call appearance it matches will indicate the same.
• The bridge appearance will only alert if the call appearance is alerting. For example, direct
intercom and paging call to the call appearance will show on the bridged appearance but will
not give any audible alert.
• If the bridged appearance user put the call on hold, the call appearance indicates 'on hold
elsewhere'.
• Bridged appearances to a user who has logged out, or has logged into a phone without
appearance buttons will not operate.
• If the bridged appearance user has 'do not disturb' (DND) enabled, the bridge appearance
button icon or lamps operates but alerting and ringing line preference selection is not applied
unless the caller is in the user’s DND exception list.
• Bridged appearance buttons are not supported between users on different systems in a
multi-site network.
Related links
Bridged Appearance Buttons on page 1166

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Bridged Appearance Buttons

Bridged Appearance Button Indication

On phones with a text display area next to the button, the name of the bridged user and the label
from the bridged user's call appearance key are displayed.
The following table shows how the different states of call appearance buttons (alerting, held,
etc) are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay on page 1188.
Icon Button Dual LED Appearance Button State
Button
Red off, Idle
Green off. The bridged appearance is not in use.
Red off, Alerting
Flashing icon. Green steady The matching call appearance is alerting for an incoming call. This is
flash. accompanied by ringing. If the user is already on a call, only a single ring
is given.
Red on, Alerting + Selected
Flashing icon. Green steady As above but Ringing Line Preference has made this the user's current
flash. selected button.
Red off, In Use Elsewhere
Green on. The matching call appearance button is in use.
Red on, In Use Here
Green on. The user has made a call or answered a call on the bridged appearance,
or bridged into it.
Red off, On Hold Here
Green fast flash. The call has been put on hold by this user.
Red off, On Hold Elsewhere
Green The call on that call appearance has been put on hold by another user.
intermittent
flash.
Icon Red off, Inaccessible
flashes off. Green broken The button pressed is not usable. The call is still dialing, ringing or cannot
flash. be bridged into.

Related links
Bridged Appearance Buttons on page 1166

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Chapter 106: Call Coverage Buttons

Call coverage allows a user to be alerted when another user has an unanswered call.

The user being covered does not necessarily have to be a key and lamp user or have any
programmed appearance buttons. Their Individual Coverage Time setting (default 10 seconds) sets
how long calls will alert at their extension before also alerting on call coverage buttons set to that
user.
The user doing the covering must have appearance buttons including a call coverage appearance
button programmed to the covered users name.
Call coverage appearance buttons are not supported between users on different systems in a
multi-site network.
Related links
Call Coverage Example 1 on page 1171
Call Coverage Example 2 on page 1172
How is Call Coverage Treated? on page 1173
Call Coverage Button Indication on page 1174

Call Coverage Example 1

In this example, the covering user is able to answer their colleagues call when it rings
unanswered. Both users have Ringing Line Preference and Auto Hold on.

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Call Coverage Buttons

Covered User Covering User Both Phones Idle

Our user has a call coverage button
to cover their colleague.

Call to Covered User

A call arrives for the covered user.

Call Alerts to Coverage

After ringing for the covered user's
Individual Coverage Time, the call
also begins alerting on the call
coverage button .
Covering User Answers
By going off hook or pressing the
alerting button, the covering user
has answered the call.

Related links
Call Coverage Buttons on page 1171

Call Coverage Example 2

In this example, the covered user has calls on all their available call appearances. Both users
have Ringing Line Preference and Auto Hold on.
Covered User Covering User Calls in Progress
The covered user already has a
number of calls in progress on all
their call appearance keys.

Table continues…

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 How is Call Coverage Treated?

Call Alerts to Coverage

The covered user is treated as
busy, so their next call goes
immediately to call coverage.

Covering User Answers

The covering user has answered
the call.

Related links
Call Coverage Buttons on page 1171

How is Call Coverage Treated?

Whose user settings control the call ?
Until answered, calls alerting on a call coverage button follow the settings of the user to
which the call was originally directed.
Once answered, the call follows the user settings of the user who answered it.
Coverage is applied to :
• Internal calls dialed to the covered user's extension number.
• External calls routed to the covered user by a incoming call route.
• Calls forwarded internally by the covered user or on follow me from the covered user.
Coverage is not applied to :
• Hunt group calls to a hunt group of which the covered user is a member.
• Calls forwarded to the covered user using forward or follow me functions.
• Calls alerting on the covered user's bridged appearance and call coverage buttons.
• Coverage is only applied to calls alerting on a line appearance if the call was also routed to
that user by an incoming call route.
• Page and intercom calls.
• Parked, transferred and held calls ringing back to the user.
• Automatic callback calls set by the covered user.
• Voicemail ringback calls.
• Call coverage appearance buttons are not supported between users on different systems in a
multi-site network.

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Call Coverage Buttons

Coverage is applied :
• If the covered user's phone is available, call coverage is applied only after the covered user's
Individual Coverage Time has expired.
• If the covered user's phone is busy, call coverage is applied immediately.
• If the covered user is using follow me or forward all to an internal number to divert their calls,
call coverage is still applied.
• If the covered user has 'do not disturb' on, call coverage is applied immediately except for
calls from numbers in the covered user's do not disturb exceptions list.
Other items :
If the call is not answered after the covered user's No Answer Time it will go to the covered user's
voicemail if available or follow their forward on no answer settings.
If the covered user has several alerting calls, the call answered by the call coverage button is the
covered user's longest ringing call.
Calls will not alert at a covering user who has 'do not disturb' enabled, except when the calling
number is in the covering user's do not disturb exception list.
Related links
Call Coverage Buttons on page 1171

Call Coverage Button Indication

On phones with a text display area next to the button, the name of the covered user is displayed
followed by the word Cover.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of call appearance buttons (alerting, held,
etc) are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay on page 1188.
Icon Button Dual LED Appearance Button State
Button
Red off, Idle
Green off. The button is not in use.
Red off, Alerting
Flashing icon. Green steady The call coverage is alerting for an unanswered call at the covered user's
flash. phone. This is accompanied by ringing. If the user is already on a call,
only a single ring is given.
Table continues…

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 Call Coverage Button Indication

Icon Button Dual LED Appearance Button State

Button
Red on, Alerting + Selected
Flashing icon. Green steady As above but Ringing Line Preference has made this the user's current
flash. selected button.
Red on, In Use Here
Green on. The user has answered the call requiring coverage.
Red off, On Hold Here
Green fast flash. The covered call has been put on hold by the call coverage button user.

Related links
Call Coverage Buttons on page 1171

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Chapter 107: Line Appearance Buttons

Line appearance buttons allow specific individual line to be used when making calls or answered
when they have an incoming call. It also allows users to bridge into calls on a particular line.

Incoming call routing is still used to determine the destination of all incoming calls. Line appearance
buttons allow a call on a specific line to alert the button user as well as the intended call destination.
When these are one and the same, the call will only alert on the line appearance but can still receive
call coverage.
When alerting on suitable phones, details of the caller and the call destination are shown during the
initial alert.
Individual line appearance ID numbers to be assigned to selected lines on a system. Line
appearance buttons are only supported for analog, E1 PRI, T1, T1 PRI, and BRI PSTN trunks;
they are not supported for other trunks including E1R2, QSIG and IP trunks.
Line appearance buttons are not supported for lines on remote systems in a multi-site network.
Using Line Appearances for Outgoing Calls
In order to use a line appearance to make outgoing calls, changes to the normal external dialing
short codes are required. For full details see Outgoing Line Programming on page 1203.
Private Lines
Special behavior is applied to calls where the user has both a line appearance for the line involved
and is also the Incoming Call Route destination of that call. Such calls will alert only on the Line
Appearance button and not on any other buttons. These calls will also not follow any forwarding.
Related links
Line Appearance Example 1 on page 1177
Line Appearance Example 2 on page 1177
How are Line Appearances Treated? on page 1178

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 Line Appearance Example 1

Line Appearance Button Indication on page 1179

Line Appearance Example 1

In this example, the user is able to answer a call alerting on a particular line.
Line Goes Active
A call is active on the line with line ID number 601. This is indicated as 'in use
elsewhere'.
For an incoming call, the line will show active but will not alert until call routing has
been determined. On analog ICLID lines, alerting is delayed until the ICLID that
might be used to do the call routing has been received.
Line Appearance Alerting
The routing of the call has been complete and it is ringing against its destination.
On our user's phone the line appearance also alerts and ringing line preference has
made it the current selected button.
Answer Call
By going off hook or pressing the line appearance, our user has answered the call
on that line.

Related links
Line Appearance Buttons on page 1176

Line Appearance Example 2

In this example, two users exchange a call using line appearance buttons set to the same line.
Note that this requires that the user who first answers the call to have Cannot be Intruded off.
Both users have Ringing Line Preference and Auto Hold on.
Idle
The two users has line
appearances for the same line.

Table continues…

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Line Appearance Buttons

Call Alerts
A call arrives. Either user can
answer it by pressing the alerting
line appearance

Call Answered
The first user has answer the call.

Line Held
The first user has put the call on
hold.

Line Retrieved
The second user has retrieved
the held call by pressing the line
appearance.

Related links
Line Appearance Buttons on page 1176

How are Line Appearances Treated?

Incoming Calls
• Until answered using a line appearance button, incoming calls alerting on a line
appearance, follow the settings of the incoming call route's destination group or user.
They do not follow the settings of any line appearance user.
• If an incoming calls destination is voicemail, or once the incoming call has passed from
its destination to voicemail, it cannot be answered or bridged into using a line appearance
button.
• If the line appearance user is also the incoming call route destination for the call, the call will
alert on their line appearance only. In this case:
- It will alert on the line appearance even if all call appearances are in use.
- The call will not follow any of the user's forwarding settings .
- The call will receive call coverage from other user's with call coverage buttons set to the
line appearance user.
- The ring delay used is that of the first free call appearance.

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 Line Appearance Button Indication

• For analog lines set to ICLID, any line appearances show active while the system waits for
ICLID information. During this time the line has not been routed and cannot be answered
using a line appearance button.
• Calls alerting on a line appearance can also alert on a call coverage appearance on the
same phone. If Ringing Line Preference is set, the current selected button will change from
the line appearance to the call coverage appearance.
• If the line appearance user has do not disturb (DND) enabled, the line appearance button
icon or lamps will still operate but alerting and ringing line preference selection are not
applied unless the caller is in their DND exception list.
Outgoing Calls
• In order to be used for making outgoing calls, some additional system programming may be
required. See Outgoing Line Programming.
• Calls made on a call appearance, which are routed out on a line for which the user also has a
line appearance, will remain on the call appearance. The line appearance will indicate 'in use
elsewhere'.
Additional Notes
• Line appearance buttons are not supported for lines on remote systems in a multi-site
network.
• Where a line appearance button is used to answer a call for which automatic call recording
is invoked, the recording will go to the automatic recording mailbox setting of the original call
destination.
• If a call indicated by a line appearance is parked, it cannot be joined or unparked by using
another line appearance.
• Calls alerting on a line appearance do not receive call coverage or go to a users voicemail
unless the user was the call's original incoming call route destination.
Related links
Line Appearance Buttons on page 1176

Line Appearance Button Indication

On phones with a text display area next to the button, the label Line and the line number are
displayed.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of call appearance buttons (alerting, held,
etc) are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay on page 1188.

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Line Appearance Buttons

Icon Button Dual LED Appearance Button State

Button
All off. Idle
The associated line is not in use.
Red on. Idle + Selected
Green off. The associated line is not in use but the button is the user currently
selected button.
Red off Alerting
Flashing icon. Green steady The line is ringing at its incoming call route destination. This is
flash. accompanied by ringing. If the user is already on a call, only a single
ring is given.
Red on Alerting + Selected
Flashing icon. Green steady As above but Ringing Line Preference has made this the user's current
flash. selected button.
Red off In Use Elsewhere
Green on. The line is in use.
Red on In Use Here
Green on. The user has answered the line, made a call on it or bridged into the call
on the line.
Red off On Hold Here
Green fast flash. The call on the line has been put on hold by this user.
Red off On Hold Elsewhere
Green The call on the line has been put on hold by another appearance button
intermittent user.
flash.
Icon Red off Inaccessible
flashes off. Green broken The button pressed is not accessible. The call is still dialing, ringing,
flash. routing or cannot be bridged into.

Related links
Line Appearance Buttons on page 1176

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Chapter 108: Appearance Button Features

Appearance functions are only supported on Avaya phones which have programmable buttons and
also support multiple calls. Appearance functions are also only supported on those buttons that
have suitable adjacent indicator lamps or a display area. Appearance buttons are not supported
across a multi-site network.
Related links
Selected Button Indication on page 1181
Idle Line Preference on page 1182
Ringing Line Preference on page 1184
Answer Pre-Select on page 1186
Auto Hold on page 1187
Ring Delay on page 1188
Delayed Ring Preference on page 1189
Collapsing Appearances on page 1191
Joining Calls on page 1192
Multiple Alerting Appearance Buttons on page 1194
Twinning on page 1195
Busy on Held on page 1195
Reserving a Call Appearance Button on page 1196
Logging Off and Hot Desking on page 1196
Applications on page 1197

Selected Button Indication

During appearance button usage, one of the user's appearance buttons may be indicated as
the user's current selected button. This is the appearance button already in use, or if idle, the
appearance button that will be used if the user goes off hook by lifting the handset.
On phones with a display area next to each button, the current selected button is indicated by
either an _ underscore of the button label, or a shaded background. On phones with twin LED
lamps, the current selected button is indicated by the red lamp being on.
The system sets which appearance button is the current selected button using the following
methods:

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Appearance Button Features

Method Description
Idle Line This feature can be set on or off for each individual user, the default is on. When on, it
Preference sets the current selected button as the first available idle call/line appearance button. See
Idle Line Preference on page 1182.
Ringing Line This feature can be set on or off for each individual user, the default is on. When on,
Preference it sets the current selected button as the button which has been alerting at the user's
phone for the longest. Ringing Line Preference overrides Idle Line Preference. See
Ringing Line Preference on page 1184.
Delayed Ring This setting is used in conjunction with ringing line preference and appearance buttons
Preference set to delayed or no ring. It sets whether ringing line preference should observe or
ignore the delayed ring applied to the user's appearance buttons when determining
which button should have current selected button status.
User Selection The phone user can override both Idle Line Preference and Ringing Line Preference
by pressing the appearance button they want to use or answer. That button will then
remain the current selected button whilst active.
If the user currently has a call connected, pressing another appearance button either
holds or disconnect that call. The action is determined by the system's Auto Hold
setting.

Answer Pre-Select
Normally when a user has multiple alerting calls, only the details of the call on current selected
button are shown. Pressing any of the alerting buttons will answer the call on that button, going
off-hook will answer the current selected button.
Enabling the user telephony setting Answer Pre-Select allows the user to press any alerting
button to make it the current selected button and displaying its call details without answering that
call. To answer a call when the user has Answer Pre-Select enabled, the user must press the
alerting button to display the call details and then either press the button again or go off-hook.
Related links
Appearance Button Features on page 1181

Idle Line Preference

Idle Line Preference determines the user's currently selected button as the first available idle
call/line appearance button. Selected button indication is applied to that button and if the user
goes off-hook, for example by lifting their handset, an outgoing call is started on that button.
• ?Why Would I Use Just Idle Line Preference In environments that are focused on making
outgoing calls, for example telemarketing, incoming calls are infrequent and user's go off-
hook expecting to make a call. Using Idle Line Preference without Ringing Line Preference
ensures that the user doesn't inadvertently answer a call when expecting to make a call.
• If all the available call/line appearance buttons are in use, no current selected button choice
is made by Idle Line Preference. In this case, going off hook will have no effect.

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 Idle Line Preference

• For appearance button users with Idle Line Preference off, going off-hook (lifting the
handset or pressing SPEAKER, HEADSET, etc) will have no effect until an appearance
button is pressed.
• By default Idle Line Preference is on for all users.
• Idle Line Preference is overridden by Ringing Line Preference if also on for the user.
Idle Line Preference Example 1
In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been programmed.
Phone Idle
The phone is idle. The current selected button determined by Idle Line Preference
is the first available idle call appearance button. This is shown by the _ underscore
of the button text.
First Call to User
A call for the user arrives. It alerts on the first available call appearance button. Idle
Line Preference has changed the current selected button to the next available idle
call appearance.
User Goes Off Hook
1. With the call still alerting, if the user goes off hook, it will be interpreted as
making a call using the currently selected button, not as answering the alerting
button.
2. To answer the alerting call, the user should press the alerting button.

Idle Line Preference Example 2

In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been not programmed.
Two Calls Alerting
The users has two incoming calls alerting. Idle Line Preference has set the
currently selected button to their third call appearance.

First Caller Abandons

If the first incoming caller disconnects, the currently selected button changes to the
first call appearance as this is now the first available idle call appearance button.

Idle Line Preference Example 3

In this example, both Idle Line Preference and Ringing Line Preference are set for the user.
Phone Idle
The phone is idle and Idle Line Preference has assigned current selected button
to the first call appearance.

Table continues…

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Appearance Button Features

Call Alerting
A call has arrived and Ringing Line Preference keeps the current selected button
at the first call appearance.

Call Answered
With the call answered it retains current selected button status.

Call Held
When the call is put on hold, Idle Line Preference assigns current selected button
status to the next available call appearance button.

Idle Line Preference Example 4

In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been programmed.
All Call Appearances Alerting
In this case, all the users call appearance buttons are alerting incoming calls. Idle
Line Preference has changed the currently selected button to the first available line
appearance.

Related links
Appearance Button Features on page 1181

Ringing Line Preference

Ringing Line Preference determines the user's currently selected button as the button which has
been alerting the longest. Selected button indication is applied to that button and if the user goes
off-hook, for example by lifting their handset, the alerting call on that button is answered.
• Ringing Line Preference includes calls alerting on call appearance, line appearance, bridged
appearance and call coverage buttons.
• Ringing Line Preference overrides Idle Line Preference.
• By default Ringing Line Preference is on for all users.
• Ringing Line Preference Order When a user's longest waiting call alerts on several of the
user's appearance buttons and Ringing Line Preference is set for the user, the order used for
current selected button assignment is;
- Call appearance.
- Bridged appearance.
- Call coverage.

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 Ringing Line Preference

- Line appearance.
• Example:
A user has a call to a covered user alerting initially on a line appearance button. Ringing Line
Preference assigns current selected button status to the line appearance. When the same
call also begins to alert on the call coverage appearance button, current selected button
status switches to the call coverage appearance button.
• Ring Delay and Ringing Line Preference
Appearance buttons can be set to Delayed Ring or No Ring. These buttons still alert visually
but do not give an audible ring or tone. Ringing line preference is still applied to alerting
buttons even if set to Delayed Ring or No Ring.
• Delayed Ring Preference
For users with Ringing Line Preference selected, their Delayed Ring Preference setting
sets whether ringing line preference is used or ignores buttons that are visually alerting but
have Delayed Ring or No Ring set. The default is off, ie. ignore ring delay.
Ringing Line Preference Example 1
In this example, both Ring Line Preference and Idle Line Preference have been set for the user.
They also have Ringing Line Preference on and Auto Hold is on. Answer Pre-Select is off.
Phone Idle
The phone is idle. The current selected button has been determined by Idle Line
Preference as the first available idle call appearance button. This is shown by the _
underscore next to that button.
First Call Alerting
A call for the user arrives. It alerts on the first available call appearance button.
Ringing Line Preference uses this as the currently selected button as it is the only
alerting call.
Second Call Alerting
Another call for the user arrives. It alerts on the next available call appearance
button. As the first call has been alerting longer, under Ringing Line Preference it
retains the currently select button status.
The First Call Abandons
The first caller disconnects. Ringing Line Preference changes the currently
selected button status to the second call appearance button.

Another Call Arrives

Another call arrives. It alerts as the first free call appearance button. However the
call at the second call appearance has been alerting longer and so under Ringing
Line Preference retain the currently selected button status.

Ringing Line Preference Example 2

In this example, the user has both Ring Line Preference and Idle Line Preference programmed.
They also have Ringing Line Preference on and Auto Hold is on. Answer Pre-Select is off.

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Appearance Button Features

First Call to User

A call for the user arrives. It alerts on the first available call appearance button.
Ringing Line Preference uses this as the currently selected button as it is the only
alerting call.
Call on Line 601
The user's Line Appearance is alerting due to an incoming call on the associated
line. Details of the call and its destination are shown. Ringing Line Preference
keeps the currently selected button status on the call appearance button as this
has been alerting longest.
Second Call to User
A second call to the user arrives and alerts on the second call appearance button.
Ringing Line Preference keeps the currently selected button status on the call
appearance button as this has been alerting longest.
The First Caller Abandons
The first call to the user disconnects. Ringing Line Preference passes the currently
selected button status to the Line Appearance button as this has been alerting
longest.

Related links
Appearance Button Features on page 1181

Answer Pre-Select
On some phones, only the details of the call alerting or connected on the current selected button
are shown. The details of calls alerting on other buttons are not shown or only shown briefly
when they are first presented and are then replaced again by the details of the call on the current
selected button.
By default, pressing any of the other alerting buttons will answer the call on that button. Answer
pre-select allows a user to press alerting buttons other than the current selected button without
actually answering them. Instead the button pressed becomes the current selected button and its
call details are displayed.
Note that using answer pre-select with a currently connected call will still either hold or end that
call in accordance with the system's Auto Hold setting.
Answer Pre-Select Example 1
Phone Idle The phone is idle. The current selected button has been determined by
Idle Line Preference as the first available idle call appearance button. This is shown
by the _ underscore next to that button.

Table continues…

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 Auto Hold

First Call Alerting A call for the user arrives. It alerts on the first available call
appearance button. Ringing Line Preference uses this as the currently selected
button as it is the only alerting call.

Second Call Alerting Another call for the user arrives. It alerts on the next
available call appearance button. As the first call has been alerting longer, under
Ringing Line Preference it retains the currently select button status.

The User Presses the Second Call Appearance Pressing the second call
appearance overrides ringing line preference and assigns current selected button
status to the button without actually answering the call. The details of the caller are
displayed.
The User Answers the Call The user can press the button again to answer the
call or just go off-hook to answer as it is now the currently selected button.

Related links
Appearance Button Features on page 1181

Auto Hold
Auto Hold is a system wide feature that affects all appearance button users. This feature
determines what happens when a user, who is already on a call, presses another appearance
button. The options are:
• If Auto Hold is off, the current call is disconnected.
• If Auto Hold is on, the current call is placed on hold.
Auto Hold Example 1
In this example, the user has two calls currently shown on call appearance buttons. Answer
Pre-Select is off.
1. This user has three call appearance buttons. They have answer one call and
are still connected to it, shown by the icon. A second call is now alerting on
their second call appearance button, shown by the icon.
2. What happens when the user presses the second call appearance key is
determined by the system's Auto Hold setting:
Table continues…

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Appearance Button Features

Auto Hold On
When the second call appearance key is pressed, that call is answered and the first
call is put on hold, shown by the icon. The user can switch between calls using
the call appearance buttons and make/receive other calls if they have additional
call appearance buttons
Auto Hold Off
When the second call appearance key is pressed, that call is answered and the first
call is disconnected.

Related links
Appearance Button Features on page 1181

Ring Delay
Ring delay can be applied to appearance buttons. This option can be used with all types of
appearance buttons and can be selected separately for each appearance button a user has. Using
ring delay does not affect the buttons visual alerting through the display and display icons or
button lamps.
Ring delay is typically used with line appearance buttons for lines which a user wants to monitor
but does not normally answer. However ring delay can be applied to any type of appearance
button.
The selectable ring delay options for an appearance button are listed below. The option is selected
as part of the normal button programming process.
Option Description
Immediate Provide audible alerting as per normal system operation.
Delayed Ring Only provide audible alerting after the system ring delay or, if set, the individual
user's ring delay.
No Ring Do not provide any audible alerting.

There are two possible sources for the delay used when delayed ringing is selected for a button.
• User > Telephony > Multi-line Options > Ring Delay: Default = Blank (Use system setting),
Range 1 to 98 seconds. This setting can be used to override the system setting. It allows a
different ring delay to be set for each user.
• System > Telephony > Telephony > Ring Delay: Default = 5 seconds, Range 1 to 98
seconds. This is the setting used for all users unless a specific value is set for an individual
user.
Notes
• Calls That Ignore Ring Delay - Ring delay is not applied to hold recall calls, park recall calls,
transfer return calls, voicemail ringback calls and automatic callback calls. For phones using

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 Delayed Ring Preference

Internal Twinning, ring delay settings are not applied to calls alerting at a secondary twinned
extension (except appearance buttons set to No Ring which are not twinned).
• Auto Connect Calls - Ring delay is applied to these calls before auto-connection. This does
not apply to page calls.
• Multiple Alerting Buttons - Where a call is presented on more than one button on a user's
phone, see Multiple Alerting Buttons, the shortest delay will be applied for all the alerting
buttons. For example, if one of the alerting buttons is set to Immediate, that will override any
alerting button set to Delayed Ring. Similarly if one of the alerting buttons is set to No Ring,
it will be overridden if the other alerting button is set to Immediate or Delayed Ring.
• Line Appearance Buttons - Calls routed to a user that could potentially be presented on
both a call appearance button and a line appearance button are only presented on the line
appearance button. In this scenario, the ring delay settings used is that of the first free call
appearance button.
• Delay on Analog Lines - Analog lines set to Loop Start ICLID already delay ringing whilst
the system waits for the full ICLID in order to resolve incoming call routing. In this scenario
the ring delay operates in parallel to the routing delay.
• Ring Delay and Ringing Line Preference - Appearance buttons can be set to Delayed
Ring or No Ring. However, ringing line preference is still applied to alerting buttons even if
set to Delayed Ring or No Ring.
• The user's Delayed Ring Preference setting is used to determine whether ringing line
preference is used with or ignores buttons that are alerting but have Delayed Ring or No
Ring set.
Ring Delay Example 1
In this example, the user has a line appearance button set but configured to no ring.
Phone Idle The phone is idle. The current selected button has been determined by
Idle Line Preference as the first available call appearance button. This is shown by
the _ underscore next to that button.

Incoming Call Alerting on the Line An incoming call arrives on the line and
begin to alert somewhere on the system. The user's line appearance button shows
this visually but doesn't ring audibly. Ringing line preference would makes the line
appearance the user's currently selected button and therefore they would answer
the line if they went off-hook.

Related links
Appearance Button Features on page 1181

Delayed Ring Preference

When a call is alerting at an idle phone, by default Ringing Line Preference sets the call as the
currently selected button and if the user then goes off-hook they will answer that call.

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Appearance Button Features

In most situations this is acceptable as the user hears ringing which informs them that there is a
call waiting to be answered. If the user wants to make a call instead, they can press another call
appearance button to go off-hook on that other button.
When ring delay is being used there can potentially be a problem if the user lifts the handset to
make a call without looking at the display. If they do this while the a call is alerting silently on a
button with ring delay, the user will actually answer the waiting call rather than get dial tone to
make a call.
Once the call alerting on a button has currently selected call status, it retains that status even if a
prior call on a button with ring delay applied comes out of its ring delay period.
Delayed Ring Preference Example 1
In this example the user has a line appearance button for a line they monitor. This line appearance
button has been set to no ring as the user occasionally need to use that line but does not normally
answer calls on that line.
Phone Idle
The phone is idle. The current selected button has been determined by Idle Line
Preference as the first available call appearance button. This is shown by the _
underscore next to that button.
Incoming Call Alerting on the Line
An incoming call arrives on the line and begin to alert somewhere on the system.
The user's line appearance button shows this visually but doesn't ring audibly.
Normally ringing line preference would make the line appearance the user's
currently selected button and therefore they would answer the line if they went
off-hook expecting to make a call.
However, because Delayed Ring Preference is on for the user, ringing line
preference is not applied and idle line preference makes their current selected
button the first call appearance. If the user were to go off-hook they would be
making a call on that call appearance.
Call Alerting for the User
A call for the user arrives. It alerts on the first available call appearance button.
Ringing line preference is applied and makes that the users currently selected
button. If the user goes off-hook now that will answer the call on the call
appearance and not the line appearance.

Delayed Ring Preference Example 2

This is similar to the previous example except that the user and the line has been configured for a
15 second ring delay. This informs the users that the line has not been answered for some reason
and allows them to answer it by just going off-hook.
Phone Idle
The phone is idle. The current selected button has been determined by Idle Line
Preference as the first available call appearance button. This is shown by the _
underscore next to that button.
Table continues…

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 Collapsing Appearances

Incoming Call Alerting on the Line

An incoming call arrives on the line and begin to alert somewhere on the system.
The user's line appearance button shows this visually but doesn't ring audibly.
Because Delayed Ring Preference is on for the user, ringing line preference is not
applied and idle line preference makes their current selected button the first call
appearance. If the user were to go off-hook they would be making a call on that call
appearance.
Call Continues Alerting
When the ring delay for the line appearance expires, if no other call has taken
ringing line preference it becomes the current selected call and will be answered if
the user goes off-hook.

Related links
Appearance Button Features on page 1181

Collapsing Appearances
This topic covers what happens when a user with several calls on different appearance buttons,
creates a conference between those calls. In this scenario, the call indication will collapse to a
single appearance button and other appearance buttons will return to idle. The exception is any
line appearance buttons involved which will show 'in use elsewhere'.
Collapsing Appearances Example 1
In this example, the user will setup a simple conference. Ringing Line Preference and Idle Line
Preference are set for the user. Auto Hold for the system is on. Answer Pre-Select is off.
Initial Call
The user has a call in progress, shown on their first call appearance button. It is
decided to conference another user into the call.

Make Conference Enquiry

Pressing the CONFERENCE button on the users phone automatically places the
current call on hold and takes the phone off hook on the next available call
appearance.
Enquiry in Progress
The other extension has answered and is invited to join a conference call. The user
presses the CONFERENCE button on their phone again.

Conference Starts/Call Appearances Collapse

The conference call has started. The call appearances have collapsed to a single
appearance.

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Appearance Button Features

Related links
Appearance Button Features on page 1181

Joining Calls
Appearance buttons can be used to "join" existing calls and create a conference call. A user can
join calls that are shown on their phone as 'in use elsewhere'.
This feature is often referred to as 'bridging into a call'. However this causes confusion with
Bridged appearance buttons and so the term should be avoided.
The ability to join calls is controlled by the following feature which can be set for each user:
• Cannot be Intruded: Default = On
If this option is set on for the user who has been in the call the longest, no other user can join
the call. If that user leaves the call, the status is taken from the next internal user who has
been in the call the longest. The exceptions are:
- Voicemail calls are treated as Cannot be Intruded at all times.
- When an external call is routed off switch by a user who then leaves the call, the Cannot
be Intruded status used is that of the user who forwarded the call off switch.
- Any call that does not involve an internal user at any stage is treated as Cannot be
Intruded on. For example:
• When an external call is routed off switch automatically using a short code in the
incoming call route.
• multi-site network calls from other systems that are routed off-switch.
• VoIP calls from a device not registered on the system.
• The Can Intrude setting is not used for joining calls using appearance buttons.
The following also apply:
Inaccessible - In addition to the use of the Cannot be Intruded setting above, a call is
inaccessible if:
• The call is still being dialed, ringing or routed.
• It is a ringback call, for example a call timing out from hold or park.
• If all the internal parties, if two or more, involved in the call have placed it on hold.
• Conferencing Resources - The ability to bridge depends on the available conferencing
resource of the system. Those resources are limited and will vary with the number of existing
parties in bridged calls and conferences. The possible amount of conferencing resource
depends on the system type and whether Conferencing Center is also installed.

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 Joining Calls

• Conference Tone - When a call is joined, all parties in the call hear the system conferencing
tones. By default this is a single tone when a party joins the call and a double-tone when a
party leaves the call. This is a system setting.
• Holding a Bridged Call - If a user puts a call they joined on hold, it is their connection
to the joined call (conference) that is put on hold. The other parties within the call remain
connected and can continue talking. This will be reflected by the button status indicators. The
user who pressed hold will show 'on hold here' on the button they used to join the call. All
other appearance users will still show 'in use here'.
• Maximum Two Analog Trunks - Only a maximum of two analog trunks can be included in a
conference call.
• Parked Calls - A Line Appearance button may indicate that a call is in progress on that line.
Such calls to be unparked using a line appearance.
Joining Example 1: Joining with a Bridged Appearance
In this example, the user joins a call using a bridged appearance button. Answer Pre-Select is
off.
User with Bridged Appearance Buttons The user has bridged appearance
buttons that match their colleagues call appearance buttons.

Call on Bridged Appearance The colleague has a call in progress on their first
call appearance. This is matched on the first bridged appearance button.

User Joins the Call Pressing the bridged appearance button will take our user off
hook and join them into their colleagues call, creating a conference call.

Joining Example 2: Joining with a Line Appearance

In this example, the user joins a call by pressing a line appearance button. Answer Pre-Select is
off.
Line Goes Active A call is active on the line with line ID number 601.
If this is an incoming call, it will show active but will not alert until its call routing
has been determined. On ICLID analog lines, alerting is delayed until the ICLID that
might be used to do that routing has been received.
Line Appearance Alerting The call routing is completed and the call is now
ringing against its target. The line appearance also begins alerting and Ringing
Line Preference has made it the current selected button.

Table continues…

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Appearance Button Features

Call Answered Alerting on the line appearance has stopped but the line is still
active. This indicates that the call has probably been answered. As our user's
phone is idle, Idle Line Preference has returned the current select button to the first
available call appearance button.
User Joins the Call Our extension user has been asked by their colleague to join
the call just answered on line 601. By pressing the line appearance button they
have joined the call on that line and created a conference call.

Related links
Appearance Button Features on page 1181

Multiple Alerting Appearance Buttons

In some scenarios, it is may be potentially possible for the same call to alert on several
appearance buttons. In this case the following apply:
• Line appearance buttons override call and bridged appearance buttons
In cases where a call on a line goes directly to the user as the incoming call route's
destination, the call will only alert on the line appearance. In this scenario the ring delay
settings used is that of the first free call appearance button.
• A call can alert both call appearance, line appearance and bridged appearance buttons
The most common example of this will be hunt group calls where the hunt group members
also have bridged call appearances to each other. In this case the button used to answer the
call will remain active whilst the other button will return to idle.
• Calls on a line/bridged appearance buttons can also alert on call coverage button
In this case alerting on the call coverage button may be delayed until the covered user's
Individual Coverage Time has expired.
• Ringing Line Preference Order
When a call alerts on several of the user's appearance buttons and Ringing Line
Preference is set for the user, the order used for current selected button assignment is:
1. Call appearance.
2. Bridged appearance.
3. Call coverage.
4. Line appearance.
Example
A user has a call to a covered user alerting initially on a line appearance button. Ringing Line
Preference will assign current selected button status to the line appearance. When the same

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 Twinning

call also begins to alert on the call coverage appearance button, current selected button status
switches to the call coverage appearance button.
Ring Delay
Where ring delays are being used, the shortest delay will be applied for all the alerting buttons. For
example, if one of the alerting buttons is set to Immediate, that will override any alerting button set
to Delayed Ring. Similarly if one of the alerting buttons is set to No Ring, it will be overridden if
the other alerting button is set to Immediate or Delayed Ring.
Related links
Appearance Button Features on page 1181

Twinning
Twinning is a mechanism that allows an user to have their calls alert at two phones. The user's
normal phone is referred to as the primary, the twinned phone as the secondary.
By default only calls alerting on the primary phone's call appearance buttons are twinned.
For internal twinning, the system supports options to allow calls alerting on other types of
appearance buttons to also alert at the secondary phone. These options are set through the
User | Twinning section of the system configuration and are Twin Bridge Appearances, Twin
Coverage Appearances and Twin Line Appearances. In all cases they are subject to the
secondary having the ability to indicate additional alerting calls.
Call alerting at the secondary phone ignoring any Ring Delay settings of the appearance button
being used at the primary phone. The only exception is buttons set to No Ring, in which case calls
are not twinned.
Related links
Appearance Button Features on page 1181

Busy on Held
For a user who has Busy on Held selected, when they have a call on hold, the system treats
them as busy to any further calls. This feature is intended primarily for analog phone extension
users. Within Manager, selecting Busy on Held for a user who also has call appearance keys will
cause a prompt offering to remove the Busy on Held selection.
Related links
Appearance Button Features on page 1181

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Appearance Button Features

Reserving a Call Appearance Button

Functions such as transferring calls using a Transfer key require the user to have at least one
available call appearance button in order to complete the outgoing call part of the process.
However, by default all call appearance button are available to receive incoming calls at all times.
Through the system configuration it is possible to reserve the user's last call appearance button for
making outgoing calls only.
1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending transfer if they
already have held calls even if they have no free call appearance button available. See Context
Sensitive Transfer on page 785.
Reserving a Call Appearance
On the User | Telephony | Multi-line Options tab, select the option Reserve Last CA.
Related links
Appearance Button Features on page 1181

Logging Off and Hot Desking

Users can be setup to log in and log out at different phones, this is called 'hot desking'. All the
users settings, including their extension number, are transferred to the phone at which the user is
logged in. This includes their key and lamp settings and appearance buttons.
This type of activity has the following effect on appearance buttons:
If logged out, or logged in at a phone that doesn't support appearance button functions:
• Bridged appearances set to the user will be inactive.
• Call coverage set to the user will still operate.
If logged in at a phone with fewer buttons than programmed for the user:
• Those buttons which are inaccessible on the logged in phone will be inactive.
• Any bridged appearances to those button from other users will be inactive.
Remote Hot Desking
Release 4.0+ supports, through the addition of license keys, users hot desking between systems
within a multi-site network. However, the use of appearance buttons (call coverage, bridged
appearance and line appearance) within a multi-site network is not supported. Therefore when a
user logs in to a remote system, any such button that they have will no longer operate. Similarly
any button that other users have with the remote user as the target will not operate.
Related links
Appearance Button Features on page 1181

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 Applications

Applications
A number of system applications can be used to make, answer and monitor calls. These
applications treat calls handled using key and lamp operation follows:
SoftConsole
This application can display multiple calls to or from a user and allow those calls to be handled
through its graphical interface.
• All calls alerting on call appearance buttons are displayed.
• Calls on line, call coverage and bridged appearance buttons are not displayed until
connected using the appropriate appearance button
• Connected and calls held here on all appearance button types are displayed.
Related links
Appearance Button Features on page 1181

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Chapter 109: Programming Appearance
Buttons

About this task

This section covers the programming of appearance buttons for users into existing system
configurations.
Appearance Functions The functions Call Appearance, Bridged Appearance, Coverage and
Line Appearance are collectively known as "appearance functions". For full details of their
operation and usage refer to the Appearance Button Operation section. The following restrictions
must be observed for the correct operation of phones.
Appearance functions programmed to buttons without suitable status lamps or icons are treated as
disabled. These buttons are enabled when the user logs in on a phone with suitable buttons in those
positions.
Line appearance buttons require line ID numbers to have been assigned, see Programming Line
Appearance Numbers. The use of line appearances to lines where incoming calls are routed using
DID (DDI) is not recommended.
How many buttons are allowed? The supported limits depend on the type of system. They are 10
for IP500 V2 systems, 20 for Server Edition and 40 for Server Edition Select. The limits are applied
as follows:
• Number of bridged appearances to the same call appearance.
• Number of line appearances to the same line.
• Number of call coverage appearances of the same covered user.
Programming Appearance Buttons Using Manager
If only button programming changes are required, the configuration changes can be merged back to
the system without requiring a reboot.
Procedure
1. Start Manager and load the current configuration from the system.
2. Locate and select the user for whom appearance buttons are required.
3. Select Button Programming.

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 The number of buttons displayed is based on the phone associated with the user when the
configuration was loaded from the system. This can be overridden by selecting Display all
buttons.
4. For the required button, click the button number and then click Edit.
5. Click the ... button.

6. From the list of options that appears, click Appearance.

7. Select the type of appearance button required.
8. Use the Action Data drop-down fields to select the required settings.
Click OK.
9. Repeat for any additional call appearance buttons required.
Click OK.
10. Repeat for any other users requiring appearance buttons.
Related links
Appearance Function System Settings on page 1200
Appearance Function User Settings on page 1200
Programming Line Appearance ID Numbers on page 1202

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Programming Appearance Buttons

Outgoing Line Programming on page 1203

Appearance Function System Settings

System settings are applied to all users and calls. The system settings that affect appearance
operation are found on the System | Telephony tabs and are:
• Auto Hold
• Conferencing Tone
• Ring Delay
• Visually Differentiate External Call
Related links
Programming Appearance Buttons on page 1198

Appearance Function User Settings

User settings are applied separately to each individual user. In addition to button programming,
the following user settings are applicable to appearance button operation:
Cannot be Intruded: Default = On. This feature controls whether other users can use their
appearance buttons to join the users call. It applies when the user is the longest present internal
party already within the call.

• Individual Coverage Time (secs): Default = 10 seconds, Range 1 to 99999 seconds.

This function sets how long the phone will ring at your extension before also alerting at any
call coverage users. This time setting should not be equal to or greater than the No Answer
Time applicable for the user.
• Ring Delay: Default = Blank (Use system setting). Range = 0 (use system setting) to 98
seconds. This setting is used when any of the user's programmed appearance buttons is
set to Delayed ringing. Calls received on that button will initially only alert visually. Audible
alerting will only occur after the ring delay has expired.
• Coverage Ring: Default = Ring. This field selects the type of ringing that should be used for
calls alerting on any the user's call coverage and bridged appearance buttons. Ring selects
normal ringing. Abbreviated Ring selects a single non-repeated ring. No Ring disables
audible ringing. Note that each button's own ring settings (Immediate, Delayed Ring or No
Ring) are still applied.
The ring used for a call alerting on a call coverage or bridged appearance button will vary
according to whether the user is currently connected to a call or not.
• If not currently on a call, the Coverage Ring setting is used.

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 Appearance Function User Settings

• If currently on a call, the quieter of the Coverage Ring and Attention Ring settings is used.
Attention Ring Setting Coverage Ring Setting
Ring Abbreviated Off
Ring Ring Abbreviated Off
Abbreviated Abbreviated Abbreviated Off

• Attention Ring: Default = Abbreviated Ring. This field selects the type of ringing that should
be used for calls alerting on appearance buttons when the user already has a connected call
on one of their appearance buttons. Ring selects normal ringing. Abbreviated Ring selects
a single ring. Note that each button's own ring settings (Immediate, Delayed Ring or No
Ring) are still applied.
• Ringing Line Preference: Default = On. For users with multiple appearance buttons. When
the user is free and has several calls alerting, ringing line preference assigns currently
selected button status to the appearance button of the longest waiting call. Ringing line
preference overrides idle line preference.
• Idle Line Preference: Default = On. For users with multiple appearance buttons. When the
user is free and has no alerting calls, idle line preference assigns the currently selected
button status to the first available appearance button.
• Delayed Ring Preference: Default = Off. This setting is used in conjunction with
appearance buttons set to delayed or no ring. It sets whether ringing line preference should
use or ignore the delayed ring settings applied to the user's appearance buttons.
When on, ringing line preference is only applied to alerting buttons on which the ring delay has
expired.
When off, ringing line preference can be applied to an alerting button even if it has delayed ring
applied.
• Answer Pre-Select: Default = Off. Normally when a user has multiple alerting calls, only the
details and functions for the call on currently selected button are shown. Pressing any of the
alerting buttons will answer the call on that button, going off-hook will answer the currently
selected button. Enabling Answer Pre-Select allows the user to press any alerting button
to make it the current selected button and displaying its call details without answering that
call until the user either presses that button again or goes off-hook. Note that when both
Answer Pre-Select and Ringing Line Preference are enabled, once current selected status
is assigned to a button through ringing line preference it is not automatically moved to any
other button.
• Reserve Last CA: Default = Off. Used for users with multiple call appearance buttons. When
selected, this option stops the user's last call appearance button from being used to receive
incoming calls. This ensures that the user always has a call appearance button available to
make an outgoing call and to initiate actions such as transfers and conferences.
1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending transfer if they
already have held calls even if they have no free call appearance button available. See Context
Sensitive Transfer.
Abbreviated Ring: This option has been replaced by the Attention Ring setting above.

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Programming Appearance Buttons

Related links
Programming Appearance Buttons on page 1198

Programming Line Appearance ID Numbers

Line appearances are supported for analog, E1 PRI, T1, T1 PRI, and BRI PSTN trunks. They are
not supported for E1R2, QSIG and IP trunks.
Note that setting and changing line settings including line appearance ID numbers requires the
system to be rebooted.
Related links
Programming Appearance Buttons on page 1198

Automatic Renumbering
About this task
Procedure
1. Select Tools | Line Renumber.
2. Select the starting number required for line numbering and click OK.
3. All lines that support Line Appearance ID will be numbered in sequence.

Manual Renumbering
About this task
Procedure
1. Start Manager and load the current configuration from the system.

2. Select Line.
3. Select the line required.
The tab through which line appearance ID numbers are set will vary depending on the type
of line. A couple of examples are shown below.
a. Analog Line
On the Line Settings tab select Line Appearance ID and enter the ID required.

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 Outgoing Line Programming

b. Basic/Primary Rate Trunks

On the Channels tab select the individual channel and click Edit. Select Line
Appearance ID and enter the required ID, then click OK. Repeat for all the channels
required.

4. Click OKand repeat for any other lines.

Outgoing Line Programming

Assigning line ID numbers to lines and associating line appearance buttons to those lines is
sufficient for answering incoming calls on those lines. However, to use line appearance buttons for
outgoing calls may require further programming.
Short Codes and Outgoing Line Appearance Calls Once a line has been seized using a line
appearance button, short code matching is still applied to the number dialed. That can include
user, system and ARS short codes.

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Programming Appearance Buttons

The short codes matching must resolve to an off-switch number suitable to be passed direct to the
line.
The final short code applied must specify a 'dial' feature. This allows call barring of specific
matching numbers to be applied using short codes set to features such as 'Busy'.
Related links
Programming Appearance Buttons on page 1198

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Part 14: SMDR Call Records

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Chapter 110: Appendix: SMDR Call
Records

The control unit is able to send SMDR (Station Message Detail Reporting) records to a specified
IP address and port. Various third-party call billing applications are able to process those records to
produce call reports.
• An SMDR record is output for each call between two parties.
• The SMDR record is output when the call between the parties ends.
• In some scenarios, for example transferred calls, multiple SMDR records are output for each
part of the call. That is, each part of the call where one of the parties involved changes. The
different parts of the call are referred to as ‘call legs’ or ‘call segments’.
• Each SMDR call record is output in a CSV format with a comma between each field.
Related links
Enabling SMDR on page 1206
SMDR Record Buffering on page 1207
Checking SMDR Generation on page 1207
SMDR Record Output on page 1207
SMDR Record Format on page 1208
Call Times in SMDR on page 1208
SMDR Fields on page 1209

Enabling SMDR
SMDR output is enabled as follows:
1. Access the system configuration using your preferred manager application.
2. Select System settings and then select the SMDR tab.
3. Use the Output drop down box to select SMDR only and enter the required IP Address
and TCP Port.
4. Adjust any other SMDR output settings if required.
5. For systems in a network of IP Offices, repeat this for all systems.

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 SMDR Record Buffering

Related links
Appendix: SMDR Call Records on page 1206

SMDR Record Buffering

The system generates a record at the end of a call or each call leg. It attempts to send the record
at the time it is generated. However, if not possible, it buffers records up to the limit set for the
system. By default that is 500 records.
• Whilst buffering, it still attempts to send a record when that new record is generated. If
successful, it will also send any buffered records.
• If the buffer limit is reached, the system deletes the oldest record each time a new record is
added. The buffer is maintained through system restarts.
Related links
Appendix: SMDR Call Records on page 1206

Checking SMDR Generation

Having enabled SMDR output, the generation of records can be view by enabling the Call trace
option Call Logging in System Monitor. Note that this causes any records displayed to be
removed from the buffer.
Related links
Appendix: SMDR Call Records on page 1206

SMDR Record Output

An SMDR record is generated at the end of each call between two devices on the system.
Devices include extensions, trunk lines (or channels on a trunk), voicemail channels, conference
channels and system tones.
• SMDR records are only produced for calls which are presented to another device or a barred
short code. For example, an internal user dialing a short code that simply changes a setting
does not produce a SMDR record.
• SMDR records are generated when each call or call leg ends. Therefore the output order of
the SMDR records does not match the call start times.
• Each record contains a Call ID :
- The Call ID starts from 1,000,000 and is reset back to that value following each system
restart.

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Appendix: SMDR Call Records

- The Call ID is increased by 1 for each subsequent new call.

- When a call moves from one device to another, separate SMDR records are output each
part of the call. Each of these records has the same Call ID.
- Each record indicates in its Continuation field whether there are further records for the
same call.
Related links
Appendix: SMDR Call Records on page 1206

SMDR Record Format

The format used for the SMDR record output is:
• Each SMDR record contains call information in a comma-separated format (CSV), that is a
byte stream of variable width fields delimited by commas (0x2C).
• Each record is terminated by carriage-return (0x0D), newline (0x0A) sequence. There is no
quoting or escaping currently defined as fields do not include ',' or 'newline' characters.
Related links
Appendix: SMDR Call Records on page 1206

Call Times in SMDR

Each SMDR record can include values for ringing time, connected time, held time and parked
time. The total duration of an SMDR record is the sum of those values.
• The time when a call is not in one of the states above, is not included in the SMDR record.
• All times are rounded up to the nearest second.
• Where announcements are being used, the connected time for a call begins either when the
call is answered or the first announcement begins.
• Each SMDR record has a Call Start Time taken from the system time. For calls being
transferred or subject to call splitting, each of the multiple SMDR records for the call has the
same Call Start time as the original call.
• The UTC Time shown at the end of the record is the time at which the SMDR record was
generated.
Related links
Appendix: SMDR Call Records on page 1206

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 SMDR Fields

SMDR Fields
The format used for the SMDR record output is:
• Each SMDR record contains call information in a comma-separated format (CSV), that is a
byte stream of variable width fields delimited by commas (0x2C).
• Each record is terminated by carriage-return (0x0D), newline (0x0A) sequence. There is no
quoting or escaping currently defined as fields do not include ',' or 'newline' characters.
Each SMDR record can contain the following fields.
• Note that time values are rounded up to the nearest second.
• Empty fields are shown if the field is not applicable to the call.
No. Field Description
1. Call Start Time The call start time in the format YYYY/MM/DD HH:MM:SS. This is based on
the system time including any DST offset.
• All records relating to the same call, that is having the same Call ID, have
the same Call Start Time.
• If the system has Call Splitting for Diverts enabled, the Call Start Time
is changed to the time the forward occurred for all records following that
stage of the call. However, the records for the externally forwarded call
retain the original Call ID.
2. Connected Time Duration of the connected part of the call in HH:MM:SS format. This does not
include ringing, held and parked time. A lost or failed call will have a duration
of 00:00:00. The total duration of a record is calculated as Connected Time
+ Ring Time + Hold Time + Park Time.
3. Ring Time Duration of the ring part of the call in seconds.
• For inbound calls, this represents the interval between the call arriving at
the switch and it being answered. It does not match the time a call rang at
an individual extension.
• For outbound calls, this indicates the interval between the call being
initiated and being answered at the remote end if supported by the trunk
type. Analog trunks are not able to detect remote answer and therefore
cannot provide a ring duration for outbound calls.
4. Caller The caller’s number. If the call originated at an extension, this is the
extension number. If the call originated externally, this is the CLI of the caller
if available, otherwise blank. For SIP trunks, the field can contain the number
plus IP address. For example, [email protected].
5. Direction The direction of the call ; I for inbound, O for outbound. This value can be
used in conjunction with the Is Internal value below to determine the call
type.
Table continues…

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Appendix: SMDR Call Records

No. Field Description

6. Called Number This is the number called by the system. For a call that is transferred, this
field shows the original called number, not the number of the party who
transferred the call.
• Internal calls – The extension, group or short code called
• Inbound calls – The target extension number for the call
• Outbound calls – The dialed digits
• Voice Mail – Calls to a user's own voicemail mailbox
7. Dialed Number For internal calls and outbound calls, this is identical to the Called Number
above. For inbound calls, this is the DDI of the incoming caller.
8. Account Code The last account code attached to the call.
9. Is Internal This field indicates whether both parties on the call are internal (1) or not
(0). Note that calls to destinations on other switches in a network are treated
as internal. This value can be used in conjunction with the Direction value
above to determine the call type as follows:

Direction Is Internal Call Type

I 0 Incoming external call.
O 1 Internal call.
O 0 Outgoing external call.
10. Call ID This is a numerical identifier, which is incremented for each unique call. If the
call has generates several SMDR records, each record has the same Call
ID. Note that the Call ID is restarted from 1,000,000 following any system
restart.
11. Continuation This value indicates if the call has any further records with the same Call ID.
It is 1 if there is a further record, otherwise 0.
12. Party1 Device The device 1 number. This is usually the call initiator though in some
scenarios, such as conferences, this may vary. If an extension/hunt group
is involved in the call, its details have priority over any trunk. That includes
remote network destinations.
Type Party Device Party Name
Internal Number E<extension number> <name>
Voicemail V<9500 + channel number> VM Channel <channel number>
Conference V<1><conference number>+<channel CO Channel <conference
number> number.channel number>
Line T<9000+line number> Line <line number>.<channel if
applicable>
Other V<8000+device number> U<device class> <device
number>.<device channel>
Unknown/Tone V8000 U1 0.0
Table continues…

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 SMDR Fields

No. Field Description

13. Party1 Name The name of the device. For an extension or agent, this is the user name
encoded in UTF-8.
14. Party2 Device The other party for the call segment. Encoded as per Party1 Device above.
For barred calls, this field shows Barred.
15. Party2 Name The other parties name. See Party1 Name above. For barred calls, this field
shows Barred.
16. Hold Time The number of seconds the call has been held during this call segment.
17. Park Time The number of seconds the call has been parked during this call segment.
18. Authorization Valid This field is used for authorization codes. This field shows 1 for valid
authorization or 0 for invalid authorization. This is Blank, no code is used.
19. Authorization Code For security, this field shows n/a regardless of the authorization code used.
This is blank, no code is used.
20. User Charged This and fields 21 to 27 are used for ISDN Advice of Charge (AoC). If blank,
AoC is not being used. This field indicates the user to which the call charge
has been assigned. This is not necessarily the user involved in the call.
21. Call Charge The total call charge calculated using the line cost per unit and user markup.
22. Currency The currency. This is a system wide setting set in the system configuration.
23. Amount at Last User The current AoC amount at user change.
Change
24. Call Units The total call units.
25. Units at Last User The current AoC units at user change.
Change
26. Cost per Unit This value is set in the system configuration against each line on which AoC
signalling is set. The values are 1/10,000th of a currency unit. For example,
if the call cost per unit is £1.07, a value of 10700 should be set on the line.
27. Mark Up Indicates the mark up value set in the system configuration for the user to
which the call is being charged. The field is in units of 1/100th, for example
an entry of 100 is a markup factor of 1 .
28. External Targeting This field indicates who or what caused the external call and a reason
Cause code. For example U FU indicates that the external call was caused by the
Forward Unconditional settings of a User.
Table continues…

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Appendix: SMDR Call Records

No. Field Description

Targeted by Reason Code
HG Hunt Group. fb Forward on Busy.
U User. fu Forward unconditional.
LINE Line. fnr Forward on No Response.
AA Auto Attendant. fdnd Forward on DND.
ICR Incoming Call Route. CfP Conference proposal (consultation) call.
RAS Remote Access Cfd Conferenced.
Service.
? Other. MT Mobile Twinning.
TW Teleworker.
XfP Transfer proposal (consultation) call.
Xfd Transferred call.
29. External Targeter ID The associated name of the targeter indicated in the External Targeting
Cause field.
• For hunt groups and users, this is their name in the system configuration.
• For an incoming call route, this is the route’s Tag value if set, otherwise
ICR.
30. External Targeted This field is used for forwarded, Incoming Call Route targeted and mobile
Number twin calls to an external line. It shows the external number called by the
system as a result of the off-switch targeting whereas other called fields give
the original number dialled.
31. Calling Party Server IP This IP address identifies the server where the calling extension is logged in.
Address
32. Unique Call ID for the Numerical value that is a unique identifier of the call on the server where the
Caller Extension call was initiated.
33. Called Party Server IP This IP address identifies the server where the called extension is logged in.
Address If the field does not contain an IP address, then the call is to a trunk outside
the IP Office network.
34. Unique Call ID for the Numerical value that is a unique identifier of the call on the server where the
Called Extension called extension is logged in.
35. SMDR Record Time The system date and time, not including any DST offset, at which the SMDR
record was generated. It uses the format YYYY/MM/DD HH:MM:SS.
36. Caller Consent This field is used for calls going through an auto-attendant service that is
Directive configured to ask for caller consent to some choice.
• 0 = Consent Not Requested
• 2 = Consent Given
• 6 = Consent Denied
Table continues…

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 SMDR Fields

No. Field Description

37. Calling Number Show the authentication level provided by the ISP on SIP lines configured to
Verification use calling number verification. Shows A, B, C or N/A is not authentication
level information provided. A record is still shown for calls which the system
rejects due to failed authentication. For more details, see SIP Calling
Number Verification (STIR/SHAKEN) on page 866.

Related links
Appendix: SMDR Call Records on page 1206

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Chapter 111: SMDR Examples

The following are examples of system SMDR records for common call scenarios.
In the following examples, the underlined fields indicate key values in the interpretation of the
scenario. ... is used to indicate that further fields have been omitted for clarity as they are not
relevant to the example.
Related links
SMDR Example: Lost Incoming Call on page 1215
SMDR Example: Transfer on page 1215
SMDR Example: Call Answered by Voicemail on page 1216
SMDR Example: Call Transferred to Voicemail on page 1216
SMDR Example: Internal Call on page 1216
SMDR Example: External Call on page 1216
SMDR Example: Outgoing Call on page 1217
SMDR Example: Voicemail Call on page 1217
SMDR Example: Parked Call on page 1217
SMDR Example: Incoming Call with Account Code on page 1218
SMDR Example: Conference Using Conference Add Short Code on page 1218
SMDR Example: Conference Using Conference Button on page 1219
SMDR Example: Adding a Party to a Conference on page 1219
SMDR Example: Busy/Number Unavailable Tone on page 1220
SMDR Example: Call Pickup on page 1220
SMDR Example: Internal Twinning on page 1220
SMDR Example: Park and Unpark on page 1221
SMDR Example: Distributed Hunt Group Call on page 1221
SMDR Example: Voicemail Supervised Transfer on page 1221
SMDR Example: Outgoing External Call on page 1222
SMDR Example: Rerouted External Call on page 1222
SMDR Example: External Forward Unconditional on page 1222
SMDR Example: Call Transferred Manually on page 1223
SMDR Example: Mobile Twinned Call Answered Internally on page 1223
SMDR Example: Mobile Twinned Call Answered at the Mobile Twin on page 1224
SMDR Example: Mobile Twinned Call Picked Up Using the Twinning Button on page 1224
SMDR Example: External Conference Party on page 1225

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 SMDR Example: Lost Incoming Call

SMDR Example: Call Routed by Incoming Call Route on page 1225

SMDR Example: Two Outgoing External Calls Transferred Together on page 1225
SMDR Example: Authorization code on page 1226
SMDR Example: Internal Network Call on page 1226
SMDR Example: Caller Consent Request on page 1226

SMDR Example: Lost Incoming Call

In this record, the Connected Time is zero and the Continuation field is 0, indicating that the call
was never connected. The Ring Time shows that it rang for 9 seconds before ending.
2014/06/28 09:28:41,00:00:00,9,8004206,I,4324,4324,,0,1000014155,0,E4324,Joe
Bloggs,T9161,LINE 5.1,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Transfer

In this example, 2126 has called 2102. The 1st record has the Continuation set a 1, indicating
that it the call has further records. The 3rd record has the same Call ID but the Party 2 Device
and Party 2 Name fields have changed, indicating that the call is now connected to a different
device. We can infer the blind transfer from the 2nd record which shows a call of zero Connected
Time between the original call destination 2102 and the final destination 2121.
2014/07/09
17:51,00:00:38,18,2126,O,2102,2102,,1,1000019,1,E2126,Extn2126,E2102,Extn2102,19,0,...
2014/07/09
17:52,00:00:00,7,2102,O,2121,2121,,1,1000020,0,E2102,Extn2102,E2121,Extn2121,0,0,...
2014/07/09
17:51,00:00:39,16,2126,O,2102,2102,,1,1000019,0,E2126,Extn2126,E2121,Extn2121,0,0,...

In this second example, extension 402 answers an external call and then transfers it to extension
403. Again the two legs of the external call have the same time/date stamp and same call ID.
2014/08/01
15:23:37,00:00:04,7,01707299900,I,4001,390664,,0,1000019,1,E402,Extn402,T9001,Line
1.1,6,0,...
2014/08/01
15:23:46,00:00:00,3,402,O,403,403,,1,1000020,0,E402,Extn402,E403,Extn403,0,0,...
2014/08/01
15:23:37,00:00:04,4,01707299900,I,4001,390664,,0,1000019,0,E403,Extn403,T9001,Line
1.1,0,0,...

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SMDR Examples

SMDR Example: Call Answered by Voicemail

In this example, 215 has made a call to 211. However, the Party2Device and Party2Name fields
show that the call was answered by voicemail.
2014/10/20 06:43:58,00:00:10,21,215,O,211,211,,I,28,0,E215,Extn215,V9051,VM Channel
1,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Call Transferred to Voicemail

In this example, the Continuation field being 1 in the first record tells us that it was not the end of
the call. The matching Call ID identifies the second record as part of the same call. The change in
Party 1 details between the two records show that the call was transferred to voicemail.
2014/06/28 09:30:57,00:00:13,7,01707392200,I,299999,299999,,0,1000014160,1,E4750,John
Smith,T9002,LINE 1.2,11,0,...
2014/06/28 09:30:57,00:00:21,0,01707392200,I,299999,299999,,0,1000014160,0,V9502,VM
Channel 2,T9002,LINE 1.2,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Internal Call

The Is Internal field being 1 indicates that this is an internal call. The Ring Time was 4 seconds
and the Connected Time was 44 seconds.
2014/06/26 10:27:44,00:00:44,4,4688,O,4207,4207,,1,1000013898,0,E4688,Joe
Bloggs,E4207,John Smith,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: External Call

The Is Internal field being 0 indicates that this is an external call. The Direction field being I
shows that it was an incoming call. The Ring Time was 7 seconds and the total Connected Time
was 5 seconds.
2014/08/01
15:14:19,00:00:05,7,01707299900,I,403,390664,,0,1000013,0,E403,Extn403,T9001,Line
1.2,0,0,...

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 SMDR Example: Outgoing Call

Related links
SMDR Examples on page 1214

SMDR Example: Outgoing Call

The combination of the Direction field being outbound and the Is Internal field being 0 show that
this was a outgoing external call. The line or channel used is indicated by the Party2 Name and
being a digital channel the Ring Time before the call was answered is also shown.
2014/06/28 08:55:02,00:08:51,9,4797,O,08000123456,08000123456,,0,1000014129,0,E4797,Joe
Bloggs,T9001,LINE 1.1,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Voicemail Call

The two records below show calls to voicemail. The first shows the Dialed Number as *17, the
default short code for voicemail access. The second shows the Dialed Number as VoiceMail,
indicating some other method such as the Message key on a phone was used to initiate the call.
2014/06/28 09:06:03,00:00:19,0,4966,O,*17,*17,,1,1000014131,0,E4966,John Smith,V9501,VM
Channel 1,0,0,...
2014/06/28 09:06:03,00:00:19,0,4966,O,VoiceMail,VoiceMail,,1,1000014134,0,E4966,John
Smith,V9501,VM Channel 1,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Parked Call

In this example the first record has a Park Time showing that the call was parked for 7 seconds.
The Continuation field indicates that the call did not end yet and there are further records. The
second record has the same Call ID and shows a change in the Party2Name, indicating that
another party unparked the call. Note also that both records share the same call start time.
2014/10/20
07:18:31,00:00:12,3,215,O,210,210,,1,1000038,1,E215,Extn215,E210,Extn210,0,7,...
2014/10/20
07:18:31,00:00:10,0,215,O,210,210,,1,1000038,0,E215,Extn215,E211,Extn211,0,0,...

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SMDR Examples

SMDR Example: Incoming Call with Account Code

Incoming call with Account Code
In this example, at some stage as the call was made or during the call, an Account Code has been
entered. During a call, another account code can be entered. The SMDR record shows the last
account code used before the record was generated.
2014/06/28
11:29:12,00:00:02,2,5002,I,1924,1924,123456789,0,1000014169,0,E1924,Extn1924,T9620,LINE
8.20,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Conference Using Conference Add Short

Code
In this example, a user conferences 2 calls. This creates 5 SMDR records; 2 initial 2–party calls
and then 3 calls connected to a system’s conference.
First 2101 has made a call and put it on hold (record 2), then made another call and put it on
hold (record 1) and then dialled the default short code *47 to conference their held calls (record
3). The records for the first two calls have the Continuation field set as 1 indicating that the calls
continued in further records.
Record 3 shows 2101 making a new call in which they dial *47, which places them and their held
calls into a conference. This is shown by the Party 2 Device and Party 2 Name details as being a
conference (100) and the conference channel used for each.
2014/07/09
17:55,00:00:03,3,2101,O,8262623#,8262623#,,0,1000024,1,E2101,Extn2101,T9002,Line
2.1,8,0,...
2014/07/09
17:54,00:00:29,7,2101,O,2121,2121,,1,1000023,1,E2101,Extn2101,E2121,Extn2121,23,0,...
2014/07/09 17:55,00:00:46,0,2101,O,*47,*47,,1,1000026,0,E2101,Extn2101,V11001,CO
Channel 100.1,0,0,...
2014/07/09
17:54,00:00:49,0,,O,71234567890,71234567890,,1,1000023,0,E2121,Extn2121,V11003,CO
Channel 100.3,0,0,...
2014/07/09 17:55,00:00:49,0,,O,8262623#,8262623#,,0,1000024,0,V11002,CO Channel
100.2,T9002,Line 2.1,0,0,...

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 SMDR Example: Conference Using Conference Button

SMDR Example: Conference Using Conference Button

In this example, an extension user answers a call and then brings in another user by using
the Conference button on their phone. Again we see records for the initial call, the conference
proposal call and then for the 3 parties in the conference that is created.
2014/07/09
15:05:41,00:00:04,3,203,O,201,201,,1,1000009,1,E203,Extn203,E201,Extn201,0,0,...
2014/07/09
15:05:26,00:00:09,3,207,O,203,203,,1,1000008,1,E207,Extn207,E203,Extn203,10,0,...
2014/07/09 15:05:41,00:00:08,0,,O,,,,1,1000009,0,E201,Extn201,V11001,CO Channel
100.1,0,0,...
2014/07/09 15:05:50,00:00:10,0,203,O,201,201,,1,1000010,0,E203,Extn203,V11002,CO
Channel 100.2,0,0,...
2014/07/09 15:05:26,00:00:10,0,207,O,203,203,,1,1000008,0,E207,Extn207,V11003,CO
Channel 100.3,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Adding a Party to a Conference

This example is a variant on that above. Having started a conference, extension 203 adds another
party.
2014/07/09
15:08:31,00:00:03,3,203,O,201,201,,1,1000014,1,E203,Extn203,E201,Extn201,0,0,...
2014/07/09
15:08:02,00:00:22,6,207,O,203,203,,1,1000013,1,E207,Extn207,E203,Extn203,9,0,...
2014/07/09 15:08:45,00:00:02,4,203,O,403,403,,0,1000016,1,E203,Extn203,E403,Libby
Franks,0,0,...
2014/07/09 15:08:02,00:00:24,0,207,O,203,203,,1,1000013,0,E207,Extn207,V11003,CO
Channel 100.3,0,0,...
2014/07/09 15:08:39,00:00:17,0,203,O,201,201,,1,1000015,0,E203,Extn203,V11002,CO
Channel 100.2,8,0,...
2014/07/09 15:08:31,00:00:26,0,,O,,,,1,1000014,0,E201,Extn201,V11001,CO Channel
100.1,0,0,...
2014/07/09 15:08:45,00:00:12,0,,O,403,403,,0,1000016,0,E403,Libby Franks,V11004,CO
Channel 100.4,0,0,...

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SMDR Examples

SMDR Example: Busy/Number Unavailable Tone

In this example, 2122 calls 2123 who is set to DND without voicemail. This results in 2122
receiving busy tone.
The records shows a call with a Connected Time of 0. The Call Number field shows 2123 as the
call target but the Party 2 Device and Party 2 Name fields show that the connection is to a virtual
device that is generating the audio tone.
2014/07/09 17:59,00:00:00,0,2122,O,2123,2123,,1,1000033,0,E2122,Extn2122,V8000,U1
0.0,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Call Pickup

The first record shows a call from 2122 to 2124 with a Connected Time of zero but a Ring Time
of 8. The Continuation field indicates that the call has further records.
The second record has the same Call ID but the Party 2 Device and Party 2 Name details show
that the call has been answered by 2121.
2014/07/09
18:00,00:00:00,8,2122,O,2124,2124,,1,1000038,1,E2122,Extn2122,E2124,Extn2124,0,0,...
2014/07/09
18:00,00:00:38,1,2122,O,2124,2124,,1,1000038,0,E2122,Extn2122,E2121,Extn2121,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Internal Twinning

The records for scenarios such as internal call forwarding or follow me indicate the rerouting in a
single record by having Caller and Called Number details that differ from the final Party 1 and
Party 2 details. Internal twinning differs is showing a call answered at the twin exactly the same as
having been answered at the primary.
203 is internally twinned to 201. Call from 207 to 203 but answer at 201.
2014/07/09
16:25:26,00:00:03,7,207,O,203,203,,1,1000037,0,E207,Extn207,E203,Extn203,0,0,...

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 SMDR Example: Park and Unpark

SMDR Example: Park and Unpark

Parking and unparking of a call at the same extension is simply shown by the Park Time field of
the SMDR record. Similarly, calls held and unheld at the same extension are shown by the Held
Time field of the SMDR record for the call. The records below however, show a call parked at one
extension and then unparked at another.
The records show a call from 207 to 203. 203 then parks the call shown by the Park Time. The
call is unparked by 201, hence the first record is indicated as continued in its Continuation field.
The matching Call ID indicates the subsequent record for the call.
2014/07/09
16:39:11,00:00:00,2,207,O,203,203,,1,1000052,1,E207,Extn207,E203,Extn203,0,4,...
2014/07/09
16:39:11,00:00:02,0,207,O,203,203,,1,1000052,0,E207,Extn207,E201,Extn201,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Distributed Hunt Group Call

An incoming call to site A is targeted to a distributed hunt group member on site B. They transfer
the call back to a hunt group member on site A.
2014/08/01
15:32:52,00:00:10,19,01707299900,I,4002,390664,,0,1000024,1,E209,Luther-209,T9001,Line
1.2,0,0,...
2014/08/01
15:33:19,00:00:00,2,209,I,403,403,,0,1000025,0,E209,Luther-209,E403,Extn403,0,0,...
2014/08/01
15:32:52,00:00:03,3,01707299900,I,4002,390664,,0,1000024,0,E403,Extn403,T9001,Line
1.2,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Voicemail Supervised Transfer

A call is routed to a voicemail module that performs a supervised transfer.
2014/08/01 16:36:04,00:00:09,0,01707299900,I,xfer,390664,,0,1000061,1,T9001,Line
1.1,V9508,VM Channel 8,0,0,...
2014/08/01 16:36:07,00:00:03,4,,I,402,402,,0,1000062,0,E402,Extn402,V8000,U12
0.8,0,0,...
2014/08/01
16:36:04,00:00:09,0,01707299900,I,402,390664,,0,1000061,0,E402,Extn402,T9001,Line
1.1,0,0,...

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SMDR Examples

Related links
SMDR Examples on page 1214

SMDR Example: Outgoing External Call

The External Targeting Cause indicates that the external call was caused by a user. The lack of
specific reason implies that it was most likely dialed. The External Targeter ID is the user name in
this example
2014/08/01 16:23:06,00:00:04,5,203,O,9416,9416,,0,1000035,0,E203,Extn203,T9005,Line
5.1,0,0,,,Extn203,,,,,,,,U,Extn203,...

Related links
SMDR Examples on page 1214

SMDR Example: Rerouted External Call

In this example, an incoming external call has been rerouted back off switch, shown by the
Party 1 fields and the Party 2 fields being external line details. The External Targeter Cause
shows that rerouting of the incoming call was done by an incoming call route (ICR). The External
Targeter ID in this case is the Tag set on the incoming call route. The External Targeted Number
is the actual external number call.
2014/08/01 08:14:27,00:00:03,5,392200,I,9416,200,,0,1000073,0,T9005,Line 5.1,T9005,Line
5.2,0,0,,,,0000.00,,0000.00,0,0,618,0.01,ICR,Main ICR,416,...

Related links
SMDR Examples on page 1214

SMDR Example: External Forward Unconditional

In this example, user 203 has a forward unconditional number set for calls. This is indicated by
the External Targeting Cause showing user and forward unconditional. The External Targeter ID
shows the source of the call being forwarded, in this example user 207. The External Targeted
Number shows the actual external number called by the system.
2014/08/01 16:22:41,00:00:02,5,207,O,203,203,,0,1000034,0,E207,Extn207,T9005,Line
5.1,0,0,,,Extn203,0000.00,,0000.00,0,0,618,1.00,U fu,Extn207,9416,...

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 SMDR Example: Call Transferred Manually

SMDR Example: Call Transferred Manually

In this example the internal user transfers a call to an external number. The External Targeting
Cause in the first record indicates that this external call is the result of a user (U) transfer proposal
(XfP) call. The Continuation field indicates that another record with the same Call ID will be
output.
The additional records are output after the transferred call is completed. The first relates to
the initial call prior. The second is the transferred call with the External Targeting Cause now
indicating user (U) transferred (Xfd).
2014/08/01 16:33:19,00:00:05,3,203,O,9416,9416,,0,1000044,1,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U XfP,Extn207,...
2014/08/01
16:33:09,00:00:02,2,207,O,203,203,,1,1000043,0,E207,Extn207,E203,Extn203,11,0,...
2014/08/01 16:33:19,00:00:04,0,207,O,9416,9416,,0,1000044,0,E207,Extn207,T9005,Line
5.1,0,0,,,Extn207,,,,,,,,U Xfd,Extn203,...

Related links
SMDR Examples on page 1214

SMDR Example: Mobile Twinned Call Answered Internally

For this example, user 203 has mobile twining enabled to the external number 9416 as their twin.
Their mobile dial delay is set to 2 seconds. The call is answered at the user's internal extension.
In this scenario the record for the external call part of twinning is output immediately the call is
answered internally. The Call Start Time for this record differs due to the user's Mobile Dial
Delay setting. The External Targeting Cause indicates the external call was the result of user
(U) mobile twinning (MT) settings. If the call had been answered before the mobile dial delay
expired, no external call and therefore no record would be produced. When the call is completed
the second record is output.
2014/08/01 16:17:59,00:00:00,7,,O,9416,9416,,0,1000028,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416,...
2014/08/01
16:17:58,00:00:07,9,207,O,203,203,,1,1000027,0,E207,Extn207,E203,Extn203,0,0,...

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SMDR Examples

SMDR Example: Mobile Twinned Call Answered at the

Mobile Twin
This is the same scenario as the example above except that the call is answered at the external
mobile twinning destination. Unlike the previous example, the external call record has a non-zero
Connected Time, showing that the call was also answered externally.
2014/08/01 16:17:04,00:00:06,9,,O,9416,9416,,0,1000026,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416,...
2014/08/01
16:17:02,00:00:06,11,207,O,203,203,,1,1000025,0,E207,Extn207,E203,Extn203,0,0,...

Related links
SMDR Examples on page 1214

SMDR Example: Mobile Twinned Call Picked Up Using the

Twinning Button
This is the same scenario as the example above, however after answering the call on the external
twinned device, the user has picked it up internally by using a twinning button. The first two
records are for the answered external call and are output when that call is picked up by the
internal extension. The third record is output when the call is ended internally.
2014/08/01
16:19:18,00:00:05,11,207,O,203,203,,1,1000029,1,E207,Extn207,E203,Extn203,0,0,...
2014/08/01 16:19:20,00:00:05,9,,O,9416,9416,,0,1000030,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416,...
2014/08/01
16:19:18,00:00:05,0,207,O,203,203,,1,1000029,0,E207,Extn207,E203,Extn203,0,0,...

Related links
SMDR Examples on page 1214

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 SMDR Example: External Conference Party

SMDR Example: External Conference Party

This is similar to internal conferencing (see examples above) but the conference setup and
progress records include External Targeting Cause codes for user (U) conference proposal (CfP)
and user (U) conferenced (Cfd).
2014/08/01 16:48:58,00:00:02,2,203,O,9416,9416,,0,1000066,1,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U CfP,Extn203,...
2014/08/01
16:48:37,00:00:04,3,203,O,207,207,,1,1000064,1,E203,Extn203,E207,Extn207,7,0,...
2014/08/01 16:49:04,00:00:08,0,203,O,9416,9416,,1,1000067,0,E203,Extn203,V11002,CO
Channel 100.2,0,0,...
2014/08/01 16:48:37,00:00:13,0,,O,,,,1,1000064,0,E207,Extn207,V11003,CO Channel
100.3,0,0,...
2014/08/01 16:48:58,00:00:13,0,,O,9416,9416,,0,1000066,0,V11001,CO Channel
100.1,T9005,Line 5.1,0,0,,,Extn203,,,,,,,,U Cfd,Extn203,...

Related links
SMDR Examples on page 1214

SMDR Example: Call Routed by Incoming Call Route

Call from external number 403 rerouted by incoming call route (ICR) for incoming line group 701
back out to 404.
2014/08/01 11:45:36,00:00:01,2,403,I,9404,,,0,1000007,0,T9001,Line 1.0,T9010,Line
10.0,0,0,0,n/a,,,,,,,,,ICR,ICR701,404,...

Related links
SMDR Examples on page 1214

SMDR Example: Two Outgoing External Calls Transferred

Together
This scenario shows an outgoing call which is then transferred to another outgoing call.
2009/02/19 11:13:26,00:00:06,0,203,O,9403,9403,,0,1000012,1,E203,Extn203,T9001,Line
1.0,8,0,0,n/a,,,,,,,,,U,Extn203,...
2009/02/19 11:13:36,00:00:02,0,203,O,8404,8404,,0,1000013,0,E203,Extn203,T9002,Line
2.0,0,0,0,n/a,,,,,,,,,U XfP,Extn203,...
2009/02/19 11:13:26,00:00:11,0,8404,I,404,,,0,1000012,0,T9002,Line 2.0,T9001,Line
1.0,0,0,0,n/a,,,,,,,,,LINE Xfd,0.1038.0 13 Alog Trunk:2,...

Related links
SMDR Examples on page 1214

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SMDR Examples

SMDR Example: Authorization code

In this example, an authorization code was used and the 0 indicates that it is invalid.
2014/02/20 11:04:59,00:00:00,0,319,O,,,,0,1000009,0,E319,Alice,V8000,U1 0.0,0,0,0,n/
a,,,,,,,,,U,Alice,...

In this example, the authorization code is valid.

2014/02/20 11:04:59,00:00:00,0,319,O,,,,0,1000009,0,E319,Alice,V8000,U1 0.0,0,0,1,n/
a,,,,,,,,,U,Alice,...

Related links
SMDR Examples on page 1214

SMDR Example: Internal Network Call

The SMDR records include fields (31 to 34) that identifying the calling and called IP Office
systems. These are useful for calls between systems in an IP Office network. This still requires
each system in the network to be configured to output its own SMDR records..
In this example, 806 on the 1st IP Office system (192.168.0.182) makes an internal call to 706
on the 2nd IP Office system (192.168.0.180). Both systems output their own SMDR record for the
same call.
Record from 1st IP Office system with the calling extension 806
2020/03/06
10:33:27,00:00:15,8,806,I,706,706,,1,1000018,0,E806,Extn806,E706,Extn706,7,0,,,,,,,,,,,,
,,192.168.0.182,1049,192.168.0.180,1087,
2020/03/06 10:33:56,0

Record from 2nd IP Office system with the called extension 706
2020/03/06
10:33:27,00:00:22,8,806,O,706,706,,1,1000004,0,E806,Extn806,E706,Extn706,0,0,,,,,,,,,,,,
,,192.168.0.182,1049,192.168.0.180,1087,
2020/03/06 10:33:56,0

Related links
SMDR Examples on page 1214

SMDR Example: Caller Consent Request

The actions in Embedded Voicemail auto-attendants and Voicemail Pro call flows can be assigned
a consent setting value. By selecting the particular action the caller can indicated their consent.
That value is indicated the SMDR record for the call.

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 SMDR Example: Caller Consent Request

Consent Not Requested

In this example, the call action used to route the call does not have a consent setting. Therefore,
the consent setting within the SMDR record remains 0.
2020/03/06 10:35:42,00:00:02,0,201,O,*99,*99,,1,1000000,1,E201,Extn201,V9511,VM Channel
11,0,0,,,,,,,,,,,,,,192.168.0.1,1002,192.168.0.1,1004,
2020/03/06 10:35:45,0
2020/03/06
10:35:42,00:00:02,2,201,O,*99,*99,,1,1000000,0,E201,Extn201,E202,Extn202,0,0,,,,,,,,,,,,
,,192.168.0.1,1002,192.168.0.1,1005,
2020/03/06 10:35:49,0

Consent Denied
In this example, the call action used to route the call is set to indicate consent denied. Therefore,
the consent setting in the SMDR record is changed to 6.
2020/03/06 10:35:54,00:00:02,0,201,O,*99,*99,,1,1000001,1,E201,Extn201,V9511,VM Channel
11,0,0,,,,,,,,,,,,,,192.168.0.1,1007,192.168.0.1,1009,
2020/03/06 10:35:56,6
2020/03/06
10:35:54,00:00:01,4,201,O,*99,*99,,1,1000001,0,E201,Extn201,E202,Extn202,0,0,,,,,,,,,,,,
,,192.168.0.1,1007,192.168.0.1,1010,
2020/03/06 10:36:00,6

Consent Given
In this example, the consent action used to route the call is set to indicate consent accepted.
Therefore, the consent setting in the SMDR record is changed to 2.
2020/03/06 10:36:08,00:00:02,0,201,O,*99,*99,,1,1000003,1,E201,Extn201,V9511,VM Channel
11,0,0,,,,,,,,,,,,,,192.168.0.1,1014,192.168.0.1,1016,
2020/03/06 10:36:09,2
2020/03/06
10:36:08,00:00:01,1,201,O,*99,*99,,1,1000003,0,E201,Extn201,E202,Extn202,0,0,,,,,,,,,,,,
,,192.168.0.1,1014,192.168.0.1,1017,
2020/03/06 10:36:11,2

Related links
SMDR Examples on page 1214

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Part 15: Miscellaneous

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Chapter 112: Overview of Data Routing

The system is a network router. In this role it can connect users on its LAN to remote services by
using WAN links and telephone trunk connections. It can also allow users to dial-in and then act as if
they were using a PC on the LAN.
As well as being a network router, the system is a telephone system. These dual roles allow it
to support a range of functions that involve traffic between the network and telephony interfaces.
These functions use internal data channels. The number of internal data channels that can be
connected from the system's LAN interface to its telephony interface at any time is restricted.
An internal data channel is a connection between the system's telephony and LAN interfaces. For
example a Voicemail connection, an internet connection or a RAS user.
Calls using a VCM channel do not use a data channel.
The number of data channels in use does not necessarily match the number of users:
• Several LAN network users, browsing the internet using the same service to an ISP would be a
single data channel.
• Several dial-in network users would each have a separate data channel.
The maximum number of data channels that can be simultaneously in use for voicemail is restricted.
These channels also require entry of an appropriate license.
The restriction depends on the type of control unit being used.
System Control Unit Internal Data Channels Maximum Data Channels for
Voicemail
Small Office Edition 18 10
IP403 18 10
IP406 V1 24 20
IP406 V2 40 20
IP412 100 30
IP500 V2 48[1] 40

1. Reduced to 44 when an IP500 V2 4-Port Expansion card is installed.

Related links
Network Address Translation (NAT) on page 1230
Dynamic Host Configuration Protocol (DHCP) on page 1230
Simple ISDN Internet Connection on page 1231

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Overview of Data Routing

ISDN Link Between IP Offices on page 1231

Using a Dedicated T1/PRI ISP Link on page 1232
Remote Access on page 1235
Creating a VoIP Link via the WAN Port Using PPP on page 1239

Network Address Translation (NAT)

NAT allows the addresses used within your LAN to be replaced by a different address when
connecting to an external service.
Typically a service provider will allocate you a single IP address to be used when connecting to
their service. NAT allows all your user's traffic to appears to be coming from that single address
without having to change any of your user's real addresses. This is useful as internally most
networks use addresses that have been reserved for public use within networks but are not valid
for routing across the internet (since the same addresses may be being used on other networks).
Also as stated it allows multiple users to use the same service simultaneously.
The use of NAT is automatically enabled if the system Service being used includes an IP address
that is not in the same domain as the its LAN1 IP address.
An exception to the above applies for systems with two LAN's, LAN1 and LAN2. For these units,
on each LAN, Enable NAT can be selected and then applied to traffic between the two LAN's.
Related links
Overview of Data Routing on page 1229

Dynamic Host Configuration Protocol (DHCP)

The system can act as a simple DHCP server. When switched on with a defaulted configuration,
the Control Unit request IP address information from a DHCP server. If it gets no response it
assumes the role of DHCP server for the LAN.
In DHCP Server mode, by default the Control Unit issues itself the address 192.168.42.1.
It allocates 200 addresses for DHCP clients, 192.168.42.1 to 19.168.42.200. This leaves
192.168.42.201 to 192.168.42.254 available for any computers that need to be allocated a fixed or
static IP address. 192.168.42.255 is not used as this is a broadcast address for the LAN.
Related links
Overview of Data Routing on page 1229

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 Simple ISDN Internet Connection

Simple ISDN Internet Connection

In this example, we want all non-local data traffic to be routed to the Internet. The Internet Service
Provider (ISP) has provided the account details required. Using the system's Network Address
Translation (NAT), a single account can be used for all users.

Select Service and add a normal service. Change the following settings and click OK.
Name: Internet
Account Name: As provided by the ISP.
Password: As provided by the ISP.
Telephone Number: As provided by the ISP.
Check Request DNS.

Select IP Route and add a new route. Change the following settings and click OK.
1. Leave the IP Address and IP Mask blank. This will then match any data traffic that isn't
matched by any other IP Route record.
2. Select the service created above as the Destination.
Alternate In the example above, a default IP Route was created which then routed all traffic to the
required Service. An alternate method to do this with system is to select Default Route within the
Service settings.
Related links
Overview of Data Routing on page 1229

ISDN Link Between IP Offices

To create a data link between two sites via ISDN configure the Control Unit as per the following
example:
At Site A on IP address 192.168.43.1
1. Create a Normal Service: The Service name can be any text and is used to identify this
particular Service. The Account Name and password are presented to the remote end,
therefore must match the User name and password configured at Site B. The Telephone
Number is the number of the remote end.
2. Create an IP Route: In the IP Address field enter the network address of the remote end,
not the IP address of the Control Unit. Under Destination select the Service created above.
3. Create a User: Under the Dial In tab tick Dial In On. This User account is used to
authenticate the connection from the Site B. Note that as the Service and User have
the same names, these two configuration forms are automatically linked and become an

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Overview of Data Routing

Intranet Service. The User password is displayed at the bottom of the Service tab as the
Incoming Password.
4. Setup RAS: Check the default RAS settings "Dial In" are available, otherwise create a
new one. If the RAS settings are given the same name as the Service and User they are
automatically linked and become a WAN Service. Ensure that the Encrypted Password
option is not checked when using a WAN Service.
5. Setup an Incoming Call Route: Check the default Incoming Call Route is available,
otherwise create a new one. If the Incoming Number is left blank, the Incoming Call Route
accepts data calls on any number. Under Destination select the RAS service created
above. The Bearer Capability should be AnyData.
At Site B on IP address 192.168.45.1
Repeat the above process but altering the details to create an route from Site B to Site A.
Related links
Overview of Data Routing on page 1229

Using a Dedicated T1/PRI ISP Link

This section shows an example of a dedicated WAN PPP link to an Internet Service Provider (ISP)
over a set of T1 or T1 PRI line channels. The ISP must support this mode of connection and will
need to provide details of the required settings. If multiple channels are to be used, then the ISP
must support Multilink PPP.
Related links
Overview of Data Routing on page 1229

Tasks for Using a Dedicated T1/PRI ISP Link

About this task
Procedure
1. Create a New WAN Service
2. Create the Virtual WAN Port
3. Create an IP Route
4. T1 PRI Trunk

Create a New WAN Service

About this task
A service is used to define connection settings such as name, password, bandwidth, etc.

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 Using a Dedicated T1/PRI ISP Link

Procedure

1. Select Service to display the existing services.

2. Click on and select WAN Service.
3. Select the Service tab.
4. In the Name field enter an appropriate name, such as “Internet”.
Note that the system will also automatically create User record and a RAS record with the
same name.
5. Enter the Account Name, Password and Telephone Number details provided by the ISP.
6. For the Firewall Profile select the firewall created previously.
7. Click the Bandwidth tab.
a. Set the Maximum No. of Channels to the maximum number of channels that the
service should use.
In this example, 12 channels were used.
b. Leave all the other records at their default values.
c. If the ISP has allocated IP address details these are entered through the IP tab.
If the IP Address and IP Mask define a different domain from the system LAN, then
NAT is automatically applied.
8. Click the IP tab.
a. In the IP Address field enter the IP address specified by the ISP.
b. In the IP Mask field enter the IP Mask specified by the ISP.
c. The settings shown are typical.
The actual settings must match those required by the ISP. For example, if Cisco
routers are being used then IPHC needs to be ticked.
9. Click the PPP tab.
Ensure that the following options are selected. Leave all other options at their default
settings.
• Multilink.
• Compression Mode: Disable.
• Callback Mode: Disable.
• Access Mode: Digital64
10. Click OK.

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Overview of Data Routing

Create the Virtual WAN Port

About this task
In this stage, a WAN port is defined that actually uses T1 or T1 ISDN trunk channels.
Procedure

1. Select WAN Port to display existing ports.

2. Click on and select WAN Port.
3. In the Name field, enter either LINEx.y where:
• LINE must be in uppercase.
• x is the line number. For a trunk card in Slot A, this will be 1. For a trunk card in Slot B,
this will be 5.
• y is the lowest numbered channel number to be used by the WAN link minus 1. For
example, if the lowest channel to be used is channel 1 then y = 1 - 1 = 0.
4. In the Speed field, enter the total combined speed of the maximum number of channels
sets in the Service.
In this example, 12 channels x 64000 bits = 76800.
5. Set the Mode to SyncPPP.
6. In the RAS Name field, select the name used for the Service.
7. Click OK.

Create an IP Route
About this task
By creating an IP route with blank IP address details, it becomes the default route for outgoing IP
traffic.
Procedure

1. Select IP Route to display existing routes.

2. Click on and select IP Route.
3. Leave the IP Address and IP Mask fields blank.
4. In the Destination field, select the WAN service.
5. Leave the Metric at default value of 1.
6. Click OK.
7. Configure the Line Channels This stage of the process differs according to the type of
trunk being used.
8. T1 Trunk Use the following for a T1 trunk.

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 Remote Access

9. Click Line to display the existing lines.

10. Double-click on the line previously entered in the WAN Port settings.
11. Check that the Channel Allocation order matches that required by the ISP.
Cisco routers typically use 1|24.
12. Select the channels to be used in the WAN PPP link and change their Channel Type to
“Clear Channel 64k”.

13. Click OK.

14. Click OKagain.
15. Send the configuration to the system and reboot.

T1 PRI Trunk
About this task
Use the following for a T1 PRI trunk.
Procedure

1. Click on Line to display the list of existing lines.

2. Double-click on the line previously entered in the WAN Port settings.
3. Check that the Channel Allocation order matches that required by the ISP.
Cisco routers typically use 1|23.
4. Select the channels to be used in the WAN PPP link and change their Admin to “Out of
Service”.
5. Click OK.
6. Click OKagain.
7. Send the configuration to the system and reboot.

Remote Access
The system support remote access for incoming data calls on trunks.

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Overview of Data Routing

To do remote access, an incoming call is passed through the following elements of the system
configuration.

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 Remote Access

• Incoming Call Route A Incoming Call

Route is used to match incoming remote access
calls and pass them to a RAS service as the
destination.

• RAS Service The RAS service defines

settings relating to the data traffic methods usable
with the call.

• User The user defines the name and

password required for the RAS service. The user
must have Dial In On enabled. An R setting on the
user's Source Numbers tab can be used to define
the ICLID from which RAS calls are accepted.

• Time Profile The user settings can specify

a time profile. The time profile then controls when
remote access is allowed.

• Firewall Profile The user settings can

specify a firewall profile. The firewall profile then
controls what traffic is allowed through the remote
access connection.
• Static NAT The system supports the use of
Static NAT records in firewall profiles. These are
used to translate external IP addresses to internal
IP addresses.

• System | LAN The system can provide

DHCP support for remote access connections
when it is set to Server or Dial in modes.
Alternatively the remote access client can use a
static IP address on the system's subnet.

• IP Route If the remote access client uses

a IP address that is from a different subnet from
the system, then a IP route record is required
for returning data. The RAS service is set as the
destination.

ISDN Remote Access Example

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Overview of Data Routing

Create a User The required details are:

• In the User tab: Enter a Name and Password. The system is case sensitive. Remember to
take care with passwords as this is a remote access link into your network.
• In the Dial In tab: Ensure that Dial In On is ticked. The Firewall Profile and Time Profile are
optional.

Create a RAS Record

In the RAS tab: Enter the same name as the user that you created earlier. Again, remember this
is case sensitive.

Create an Incoming Call Route

• Set the Bearer Capability to Any Data.
• In the Destination drop-down list, select the RAS record created above.
• The values that you enter for any of the other fields will depend on whether the remote user
will be calling in on a particular line, number or from a set ICLID.
1. Is a Return IP Route Needed ? Go to Step 5.

2. Create a IP Route (Optional) If the remote user has an IP address that is not in
the same domain as the system, then an IP Route is needed for return data. This is not
necessary if the remote user's dial-up connection method is set to 'Obtain an IP Address
Automatically' and the system's DHCP mode is set to Server or Dial In.
• Enter the IP Address and IP Mask of the remote system.
• In the Destination drop-down list select the RAS record created above.
Analog Remote Access Example

Configuration for a connection from an analog modem call is very similar to the ISDN example.
However the system must be able to answer modem calls. This can be done in the following ways:
• Analog Trunk Modem Mode On systems with an analog trunk card in the control unit, the
first analog trunk can be set to answer V.32 modem calls. This is done by checking the
Modem Enabled option on the analog line settings or using the default short code *9000* to
toggle this service on or off.
• IP500 ATM4 Uni Trunk Card Modem Support It is not required to switch the card's modem
port on/off. The trunk card's V32 modem function can be accessed simply by routing a
modem call to the RAS service's extension number. The modem call does not have to use
the first analog trunk, instead the port remains available for voice calls.
When using an analog modem, the Bearer Capability of the incoming call route used should be
Any Voice.

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 Creating a VoIP Link via the WAN Port Using PPP

Related links
Overview of Data Routing on page 1229

Creating a VoIP Link via the WAN Port Using PPP

A VoIP link across a leased line requires the Control Unit at both ends to have a Voice
Compression Module installed. These provide for a fixed number of channels to use VoIP at
any time. They are used to compress voice down to either 6k3 (G.723) or 8k (G.729) and provide
echo cancellation.
Both ends must using the same version of software and configured to use the same speed and
compression.
For example, consider this VoIP link:
• Site A on IP address 192.168.42.1
• Site B on IP address 192.168.45.1
Perform the following steps, once for Site A and once for Site B.
1. Create a Normal Service: The Account Name and password is presented to the remote
end, therefore must match the User name and password configured at Site B. The
Encrypted Password option can only be used if the remote end also supports CHAP.
2. Create a User: Under the Dial In tab tick Dial In On. This User account is used to
authenticate the connection from the Site B. As the Service and User have the same
name these two configuration forms are automatically linked and become an Intranet
Service. The User password is displayed at the bottom of the Service tab as the Incoming
Password.
3. Name: SiteB
4. Dial In | Dial In On: Enabled.
5. Create a RAS service: If CHAP is to be used on this link, then the Encrypted Password
option must be checked in the Service and in the RAS service. The name of the RAS
service must match the name of the Service at Site B. If the RAS service is given the same
name as the Service and User, they are automatically linked and become a WAN Service.
Ensure that the Encrypted Password option is not checked when using a WAN Service.
6. Edit the WANPort: Note - do not create a new WANPort, this is automatically detected. If
a WANPort is not displayed, connect the WAN cable, reboot the Control Unit and receive
the configuration. The WANPort configuration form should now be added.
RAS Name: SiteA
7. Create an IP Route: The IP Address is the network address of the remote end. Under
Destination select the Service created above.
8. Create a new Line: The Line Number and Line Group ID must be unique, in other words,
not used by any other line. The Gateway IP Address is the IP Address of the Control Unit

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Overview of Data Routing

at the remote end. The Compression Mode used is dependent on the Voice Compression
Card the Control Unit is running and the speed of the link.
9. Create a Short Code: To route all calls where the number dialed starts with 8 via Line
Group ID 1, therefore via the VPN Line created above.
10. Short Code: 8N
11. Telephone Number: N
12. Line Group ID: 1
13. Feature: Dial
Related links
Overview of Data Routing on page 1229

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Part 16: Further Help

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Chapter 113: Additional Help and
Documentation

The following pages provide sources for additional help.

Related links
Additional Manuals and User Guides on page 1242
Getting Help on page 1242
Finding an Avaya Business Partner on page 1243
Additional IP Office resources on page 1243
Training on page 1244

Additional Manuals and User Guides

The Avaya Documentation Center website contains user guides and manuals for Avaya products
including IP Office.
• For a listing of the current IP Office manuals and user guides, look at the Avaya IP Office™
Platform Manuals and User Guides document.
• The Avaya IP Office Knowledgebase and Avaya Support websites also provide access to the
IP Office technical manuals and users guides.
- Note that where possible these sites redirect users to the version of the document hosted
by the Avaya Documentation Center.
For other types of documents and other resources, visit the various Avaya websites (see
Additional IP Office resources on page 1243).
Related links
Additional Help and Documentation on page 1242

Getting Help
Avaya sells IP Office through accredited business partners. Those business partners provide
direct support to their customers and can escalate issues to Avaya when necessary.

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 Finding an Avaya Business Partner

If your IP Office system currently does not have an Avaya business partner providing support and
maintenance for it, you can use the Avaya Partner Locator tool to find a business partner. See
Finding an Avaya Business Partner on page 1243.
Related links
Additional Help and Documentation on page 1242

Finding an Avaya Business Partner

If your IP Office system currently does not have an Avaya business partner providing support and
maintenance for it, you can use the Avaya Partner Locator tool to find a business partner.
Procedure
1. Using a browser, go to the Avaya Website at https://www.avaya.com
2. Select Partners and then Find a Partner.
3. Enter your location information.
4. For IP Office business partners, using the Filter, select Small/Medium Business.
Related links
Additional Help and Documentation on page 1242

Additional IP Office resources

In addition to the documentation website (see Additional Manuals and User Guides on
page 1242), there are a range of website that provide information about Avaya products and
services including IP Office.
• Avaya Website (https://www.avaya.com)
This is the official Avaya website. The front page also provides access to individual Avaya
websites for different regions and countries.
• Avaya Sales & Partner Portal (https://sales.avaya.com)
This is the official website for all Avaya business partners. The site requires registration for
a user name and password. Once accessed, you can customize the portal to show specific
products and information type that you want to see.
• Avaya IP Office Knowledgebase (https://ipofficekb.avaya.com)
This site provides access to an online, regularly updated version of IP Office user guides and
technical manual.
• Avaya Support (https://support.avaya.com)

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Additional Help and Documentation

This site provide access to Avaya product software, documentation and other services for
Avaya product installers and maintainers.
- Avaya Support Forums (https://support.avaya.com/forums/index.php)
This site provides forums for discussing product issues.
• International Avaya User Group (https://www.iuag.org)
This is the organization for Avaya customers. It provides discussion groups and forums.
• Avaya DevConnect (https://www.devconnectprogram.com/)
This site provides details on APIs and SDKs for Avaya products, including IP Office. The site
also provides application notes for third-party non-Avaya products that interoperate with IP
Office using those APIs and SDKs.
• Avaya Learning (https://www.avaya-learning.com/)
This site provides access to training courses and accreditation programs for Avaya products.
Related links
Additional Help and Documentation on page 1242

Training
Avaya training and credentials ensure our Business Partners have the capabilities and skills to
successfully sell, implement, and support Avaya solutions and exceed customer expectations. The
following credentials are available:
• Avaya Certified Sales Specialist (APSS)
• Avaya Implementation Professional Specialist (AIPS)
• Avaya Certified Support Specialist (ACSS)
Credential maps are available on the Avaya Learning website.
Related links
Additional Help and Documentation on page 1242

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Index
Numerics announcements ................................................................. 472
APIs ................................................................................. 1243
4400/6400 ..........................................................................467 Apple
911 View ............................................................................ 653 push notifications ................................................731, 732
911–View ..........................................................................1103 Application Notes .............................................................1243
ARS configuration fields .................................................... 593
Attendant
A Consent ...................................................................... 883
AA Number ........................................................................ 885 Dial By Conference .....................................................892
account code configuration ................................................722 Dial By Name ..............................................................893
account code configuration fields .............................. 547, 548 Dial By Number .......................................................... 895
voice recording ........................................................... 548 Leave Message .......................................................... 896
ACO Domain Name ........................................................... 292 Park & Page ............................................................... 898
ACO line ............................................................................ 292 Replay Menu .............................................................. 900
ACO Proxy Address ...........................................................292 Speak By Name ..........................................................901
ACO T38 Fax .....................................................................296 Speak By Number ...................................................... 902
ACO VoIP .......................................................................... 293 Supervised Transfer ................................................... 897
Acquire Transfer to Auto Attendant ..........................................904
Button ....................................................................... 1078 Unsupervised Transfer ............................................... 903
Action Audio Output ......................................................................885
Dial By Conference .....................................................892 audit trail ............................................................................ 101
Dial By Name ..............................................................893 authorization codes configuration fields .............................578
Dial By Number .......................................................... 895 auto attendant configuration fields .....................565, 566, 571
Leave Message .......................................................... 896 actions ................................................................ 568, 575
Park & Page ............................................................... 898 auto intercom deny off ....................................................... 957
Replay Menu .............................................................. 900 auto intercom deny on ....................................................... 957
Speak By Name ..........................................................901 Auto-attendant
Speak By Number ...................................................... 902 Actions ........................................................................892
Supervised Transfer ................................................... 897 Prompts ...................................................................... 906
Transfer to Auto Attendant ..........................................904 Auto-Attendant ...................................................................880
Unsupervised Transfer ............................................... 903 Actions ........................................................................889
Actions ............................................................................... 892 Consent ...................................................................... 883
Ad-Hoc conference Dial By Conference .....................................................892
Add ............................................................................. 921 Dial By Name ..............................................................893
Ad-Hoc Conference ........................................................... 921 Dial By Number .......................................................... 895
Add External calls .............................................................. 910
System Conference .................................................... 927 Fallback Action ........................................................... 889
Administrator ....................................................................1242 Fallback options ..........................................................883
advanced ............................................................................. 96 Greeting ......................................................................885
erase configuration ....................................................... 96 Internal call ................................................................. 910
reboot ........................................................................... 97 Language ....................................................................881
system shutdown ..........................................................97 Leave Message .......................................................... 896
upgrade .......................................................... 98 Menu ...........................................................................885
changing the .bin file directory .............. 100 Name prompts ............................................................ 908
searching for systems ........................... 100 Park & Page ............................................................... 898
advanced view ................................................................... 114 Pre-recorded prompts .................................................907
alarms ................................................................................ 266 Recording prompts ..................................................... 906
analog extension ................................................................419 Replay Menu .............................................................. 900
analog line ......................................................................... 297 Route calls to ..............................................................910
analog options ............................................................ 299 Settings .......................................................................885
line settings .................................................................298 Short code .................................................................. 910
Announcement Speak By Name ..........................................................901
Auto-Attendant ............................................................885 Speak By Number ...................................................... 902

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Auto-Attendant (continued) certificate support (continued)
Supervised Transfer ................................................... 897 file naming and format ................................................ 643
Transfer to Auto Attendant ..........................................904 identity certificate ........................................................644
Unsupervised Transfer ............................................... 903 signing certificate ........................................................647
Auto–Attendant trusted certificate store ............................................... 646
Settings .......................................................................885 certificates ..........................................................................135
autoconnect ........................................................................511 change mode ..................................................................... 101
Avaya Push Notification ..................................................... 289 change working directory .....................................................79
close configuration ...............................................................77
cloud .................................................................................. 731
B authorization ............................................................... 731
Backup ...............................................................................607 Codec renegotiation ...........................................................398
backup/restore ..................................................................... 82 codec selection .................................................................. 857
bandwidth .......................................................................... 507 COM .................................................................................. 607
barred calls ........................................................................ 703 conference ......................................................................... 590
applying ...................................................................... 703 Conference
overriding ....................................................................704 Add a system conference ........................................... 927
Barring Calls ...................................................................... 720 Auto-Attendant ............................................................885
blind transfer ...................................................................... 782 Deleting a system conference .................................... 928
BOOTP record ................................................................... 200 Direct By Conference ................................................. 885
Break out ........................................................................... 760 Editing a system conference ...................................... 928
BRI line .............................................................................. 305 Personal Meet-Me PIN ............................................... 923
channels ..................................................................... 310 System Conference settings .......................................929
business partner locator .................................................. 1243 System Conferences .................................................. 927
busy on held validation .......................................................118 User controls .............................................................. 914
Button Conference ID ....................................................................916
911 View ..................................................................... 653 Conferencing ..................................................................... 913
Call Steal .................................................................. 1078 Capacity ......................................................................915
Emergency View .........................................................653 configuration field
button programming ...........................................................466 Access Control Lists ................................................... 282
user .............................................................................476 account code
Button programming ........................................................ 1042 voice recording ......................................548
ACO line ..................................................................... 292
ACO T38 Fax ..............................................................296
C ACO VoIP ................................................................... 293
analog .........................................................................419
Call
analog line .................................................................. 297
Auto-Attendant ............................................................910
analog line options ......................................................299
Call Barring ........................................................................ 720
analog line settings .....................................................298
call log ........................................................................255, 459
ARS ............................................................................ 593
Call Management
authorization codes .................................................... 578
Conference ................................................................. 929
auto attendant
call recordings
actions ...........................................568, 575
user playback ............................................................. 476
BOOTP ....................................................................... 200
Call Records .................................................................... 1206
BRI line ............................................................... 305, 310
call reporting ...................................................................... 476
call details ...................................................................379
call settings ........................................................................ 450
call log ........................................................................ 255
call steal .............................................................................794
control unit .................................................................. 413
Call Steal
customer call reporter .................................................276
Button ....................................................................... 1078
DHCP pools ................................................................228
Callflow .............................................................................. 925
dialer ...........................................................................282
Calling Number Verification ....................... 866, 868–870, 873
directory ......................................................................535
centralized licensing .......................................................... 672
directory services ................................................259, 263
certificate management ..................................................... 638
display
overview ..................................................................... 638
server edition mode .............................. 197
Windows certificate store ............................................640
standard mode ...................................... 196
certificate support .............................................................. 643
DNS ............................................................................ 230
file import .................................................................... 649

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WAN port (continued) WAN port (continued)
E1 line .........................................................................344 service (continued)
E1 line channels ......................................................... 350 bandwidth ..............................................507
E1 R2 Advanced .........................................................356 dial in .....................................................515
E1 R2 channels .......................................................... 354 fallback ..................................................514
E1 R2 line ................................................................... 352 IP ...........................................................510
E1 R2 MFC group .......................................................356 PPP .......................................................512
extension ............ 415, 416, 419, 421, 425, 429, 431, 432 quota ..................................................... 511
firewall profile SSL VPN .......................................515, 516
custom .................................................. 541 SSL VPN fallback ..................................518
standard ................................................ 539 SSL VPN NAPT .................................... 517
static NAT ..............................................543 SSL VPN session ..................................516
group short code ...................................................................503
fallback ..................................................490 shorty codes ............................................................... 448
overflow .................................................488 SIP advanced ............................................................. 392
select members .....................................484 SIP credentials ................................................... 342, 391
SIP ........................................................ 502 SIP DECT base .................................................. 332, 432
voice recording ......................................498 SIP DECT VoIP .......................................................... 333
voicemail ............................................... 493 SIP engineering .................................................. 341, 398
H.323 line ....................................................................311 SIP line ....................................................................... 372
H.323 short codes ...................................................... 313 SIP T38 Fax ................................................................390
H.323 VoIP ................................................................. 421 SIP transport ...............................................................376
H.323 VoIP settings .................................................... 314 SIP VoIP ............................................................. 386, 425
incoming call route SM line ........................................................................400
standard ................................................ 523 SM line T38 Fax ......................................................... 407
voice recording ..............................527, 528 SM line VoIP ............................................................... 403
IP DECT ..................................................................... 431 SMDR ......................................................................... 272
IP DECT gateway ....................................................... 317 SMTP ..........................................................................271
IP DECT line ...............................................................317 subscription ........................................................ 550, 604
IP DECT VoIP .............................................................320 system ........................................................................ 203
IP Office line ............................................................... 322 system events .............................................................265
IP Office line short codes ............................................327 system events alarms .................................................266
IP Office line T38 Fax ................................................. 330 system events configuration ....................................... 265
IP Office line VoIP settings ......................................... 327 System Manager ................................................ 253, 254
IP route T1 channels ................................................................ 360
RIP dynamic routing ..............................545 T1 line .........................................................................358
LAN settings ............................................................... 213 T1 PRI line ..................................................................363
LAN VoIP .................................................................... 215 T38 fax ........................................................................429
LAN1 ...........................................................................213 telephony .................................................................... 239
LAN2 ...........................................................................230 time profile .......................................................... 537, 667
Legacy SIP DECT line ................................................331 tones and music ......................................................... 249
license TUI ..............................................................................256
remote server ........................................555 tunnel
line .............................................................................. 290 IP security tunnel .................... 562
location .........................................................598 IKE policies ......................563
address ................................................. 601 IPSec policies .................. 564
MS Teams line ............................................................ 334 main .................................562
MS Teams line VoIP ................................................... 337 L2TP tunnel ...................................559, 561
network topology ........................................................ 223 user ...................................................... 433, 448
park and page .............................................................248 announcements .................................... 472
PRI trunks ...................................................................343 button programming ..............................466
remote access server dial in .....................................................465
PPP .......................................................519 DND ...................................................... 448
ring tones ....................................................................253 forwarding ............................................. 461
S0 line .........................................................................408 hunt group membership ........................ 472
service ..................................................505, 506 menu programming ...............................466
autoconnect ...........................................511 mobility ..................................................468

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WAN port (continued) directory configuration fields ...................................... 259, 535
user (continued) directory services
personal directory ................................. 474 HTTP .......................................................................... 263
SIP ........................................................ 474 LDAP .......................................................................... 259
source numbers .................................... 449 discovery ..............................................................................88
telephony .............................................. 450 DNS ................................................................................... 230
voice recording ......................................465 Subscription ................................................................609
voicemail ............................................... 442 do not disturb ..................................................................... 448
user rights download
forwarding ............................................. 588 recordings ...................................................................476
telephony .............................................. 581 downloading
user ....................................................... 579 manager lite ..................................................................45
user rights membership ................ 586, 587 DST ....................................................................................664
voice compression modules ....................................... 274 DTag .................................................................................. 878
voicemail .....................................................................232 DTMF .................................................................................858
VoIP .................................................................... 277, 421
VoIP security ...............................................................279
WAN port
E
advanced .............................................. 533 E1 line ................................................................................344
DLCI ......................................................532 channels ..................................................................... 350
frame relay ............................................ 531 E1 R2 line .......................................................................... 352
connect to ...........................................................................116 Advanced ....................................................................356
Consent ............................................................................. 883 channels ..................................................................... 354
Contact Center ...................................................................284 MFC group ..................................................................356
control unit ......................................................................... 413 Edit
courses ............................................................................ 1243 Daylight Saving Time ..................................................664
create new config .................................................................94 System Conference .................................................... 928
CTI Email alarm ........................................................................654
Subscription ................................................................606 embedded call reporting .................................................... 476
customer call reporter ........................................................ 276 embedded file management .............................................. 103
Customer Operations Manager ......................................... 607 embedded file management menu .................................... 123
Emergency Call
D System Alarm ............................................................. 654
Emergency View ...................................................... 653, 1103
Date ................................................................................... 663 Enable local recording ....................................................... 906
Manual ........................................................................666 Enable Local Recording .....................................................885
System Status ............................................................ 665 erase configuration .............................................................. 96
Daylight Saving Time ......................................................... 664 error pane .....................................................................59, 114
Default automatic validation settings ........................................ 59
TTS language ............................................................. 882 revalidating settings ......................................................60
Delete viewing error ................................................................. 60
System Conference .................................................... 928 exit ....................................................................................... 82
Destination export user ......................................................................... 116
Auto-attendant ............................................................ 910 Extension
details pane ..................................................................57, 114 Emergency Call Indication ..........................................653
details toolbar ...................................................................... 54 extension configuration fields ............................................ 415
DHCP pools ....................................................................... 228 analog .........................................................................419
Dial By Conference ............................................................892 extension .................................................................... 416
Dial By Name .....................................................................893 H.323 VoIP ................................................................. 421
Recording name prompts ........................................... 908 IP DECT ..................................................................... 431
Dial By Number ..................................................................895 SIP DECT base .......................................................... 432
dial in ................................................................................. 465 SIP VoIP ..................................................................... 425
Dial To Record Greeting .................................................... 885 T38 fax ........................................................................429
dialer .................................................................................. 282 VoIP ............................................................................ 421
Direct By Conference .........................................................885 extension renumber ........................................................... 115
Direct By Number .............................................................. 885
directories ............................................................................ 86

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F Headers (continued)
SIP ..............................................................................878
fallback .......................................................................490, 514 Help ................................................................................. 1242
Fallback ............................................................................. 883 Hold Reminders ................................................................. 863
Fallback Action .................................................................. 889 hold scenarios ....................................................................859
fax over SIP ....................................................................... 859 Hot desking ........................................................................760
file menu .............................................................................. 76 HTTP directory services .................................................... 263
change working directory ..............................................79 hunt group membership .....................................................472
close configuration ........................................................77 huntgroup ...........................................................................467
open configuration ........................................................ 76
save configuration ........................................................ 77
save configuration as ....................................................78
I
firewall ................................................................................539 icons
custom ........................................................................ 541 changing size ................................................................63
standard ......................................................................539 import templates .................................................................119
static NAT ................................................................... 543 import/export ........................................................................82
firewall profile configuration fields ......................................539 incoming call
custom ........................................................................ 541 call scenarios ..............................................................853
standard ......................................................................539 Incoming call route
static NAT ................................................................... 543 Auto-attendant ............................................................ 910
format SD card ...................................................................103 incoming call route configuration fields ..............................521
forums ..............................................................................1243 standard ......................................................................523
forwarding .......................................................................... 461 voice recording ................................................... 527, 528
incoming calls
G media path connection ............................................... 851
initial configuration ............................................................. 108
general security fields ........................................................ 142 Internal call
getting started ...................................................................... 42 Auto-attendant ............................................................ 910
Gmail integration ................................................................715 Intrusion ............................................................................. 716
Google TTS ....................................................................... 882 IP DECT .............................................................................431
Greeting ............................................................................. 885 IP DECT gateway .............................................................. 317
Pre-recorded ...............................................................907 IP DECT line ...................................................................... 317
Short code .................................................................. 906 IP DECT VoIP .................................................................... 320
TTS .............................................................................908 IP Office line .......................................................................322
group configuration fields ...........................................480, 485 short codes ................................................................. 327
fallback ....................................................................... 490 VoIP settings ...............................................................327
overflow ...................................................................... 488 IP Office line T38 Fax ........................................................ 330
select members .......................................................... 484 IP route
SIP ..............................................................................502 configuring .................................................................. 622
voice recording ........................................................... 498 IP route configuration fields ............................................... 544
voicemail .....................................................................493 RIP dynamic routing ................................................... 545
group operation ..................................................................765 IP security tunnel ............................................................... 562
group pane ................................................................... 55, 114 IKE policies .................................................................563
add record .................................................................... 56 IPSec policies ............................................................. 564
columns displayed ........................................................ 56 main ............................................................................562
delete record .................................................................56
show in groups ............................................................. 57
sorting ...........................................................................55
J
validate record .............................................................. 57 Java ..................................................................................... 42

H K
H.323 extension VoIP ........................................................ 421 Keepalives ......................................................................... 398
H.323 line ........................................................................... 311
short codes ................................................................. 313
VoIP setting ................................................................ 314
Headers

June 2024 Administering Avaya IP Office using Manager 1249

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L line configuration fields (continued)
IP Office line VoIP settings ......................................... 327
L2TP tunnel ....................................................................... 559 Legacy SIP DECT line ................................................331
L2TP ........................................................................... 561 MS Teams line ............................................................ 334
PPP ............................................................................ 561 MS Teams line VoIP ................................................... 337
LAN settings ...................................................................... 213 PRI trunks ...................................................................343
LAN VoIP ........................................................................... 215 S0 line .........................................................................408
LAN1 ..................................................................................213 SIP advanced ............................................................. 392
LAN2 ..................................................................................230 SIP credentials ................................................... 342, 391
Language SIP DECT base .......................................................... 332
Auto-Attendant ............................................................881 SIP DECT VoIP .......................................................... 333
TTS default ................................................................. 882 SIP engineering .................................................. 341, 398
launch Voicemail Pro ......................................................... 107 SIP line ....................................................................... 372
LDAP ................................................................................. 734 SIP T38 Fax ................................................................390
LDAP directory services .................................................... 259 SIP transport ...............................................................376
LDAP synchronization SIP URI .......................................................................379
creating a user provisioning rule .................................735 SIP VoIP ..................................................................... 386
performing .................................................................. 734 SM line ........................................................................400
Leave Message ................................................................. 896 SM line T38 Fax ......................................................... 407
Legacy SIP DECT line ....................................................... 331 SM line VoIP ............................................................... 403
SIP DECT base .......................................................... 332 T1 channels ................................................................ 360
SIP DECT VoIP .......................................................... 333 T1 line .........................................................................358
license T1 PRI line ..................................................................363
configuring .................................................................. 670 line renumber ..................................................................... 115
license configuration fields .................................................552 Local recording .................................................................. 906
remote server ............................................................. 555 Local Recording .................................................................885
license file location .............................................................................. 598
uploading .................................................................... 679 address .......................................................................601
license migration ........................................................120, 684 Location ............................................................................. 617
licensing location configuration fields ............................................... 598
enterprise branch ........................................................682 Locations ........................................................................... 664
line Log files ............................................................................. 607
MS Teams line ............................................................ 334 Loop Count ........................................................................ 885
line configuration fields ...................................................... 290 LVM greeting utility .............................................................107
ACO line ..................................................................... 292
ACO T38 Fax ..............................................................296
ACO VoIP ................................................................... 293
M
analog line .................................................................. 297 madn ................................................................................ 1123
analog line options ......................................................299 main toolbar ......................................................................... 53
analog line settings .....................................................298 Manager
BRI line ............................................................... 305, 310 synchronize passwords .............................................. 804
control unit .................................................................. 413 Time ............................................................................663
E1 line .........................................................................344 manager lite
E1 line channels ......................................................... 350 download ...................................................................... 45
E1 R2 Advanced .........................................................356 manager modes ...................................................................35
E1 R2 channels .......................................................... 354 Manuals ........................................................................... 1242
E1 R2 line ................................................................... 352 Maximum Inactivity ............................................................ 885
E1 R2 MFC group .......................................................356 Media Manager
H.323 line ....................................................................311 Subscription ................................................................606
H.323 short codes ...................................................... 313 media path connection .......................................................851
H.323 VoIP setting ......................................................314 memory card command ..................................................... 106
IP DECT gateway ....................................................... 317 Menu Announcement .........................................................885
IP DECT line ...............................................................317 menu bar commands ........................................................... 75
IP DECT VoIP .............................................................320 advanced ..........................................96–98, 100
IP Office line ............................................................... 322 audit trail ............................................... 101
IP Office line short codes ............................................327 change mode ........................................ 101
IP Office line T38 Fax ................................................. 330 embedded file management ................. 103

June 2024 Administering Avaya IP Office using Manager 1250

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tools (continued) navigation pane ............................................................54, 114
advanced (continued) navigation toolbar ................................................................ 54
erase security settings .......................... 102 network table ....................................................................... 66
format SD card ......................................103 network topology ................................................................223
launch Voicemail Pro ............................ 107 new in this release ............................................................... 33
LVM greeting utility ................................107 No Match Prompt ...............................................................885
memory card command ........................ 106 No User ..............................................................................729
recreate SD card ...................................105 NoCallerId alarm
security settings .................................... 102 suppressing ................................................................ 730
System Status .......................................107 NoUser
VM locales .............................................112 Source Numbers .........................................................797
backup/restore ..............................................................82 NTP ....................................................................................663
embedded file management ....................................... 123 Number
exit ................................................................................ 82 Auto-Attendant ............................................................885
file ....................... 76, 82, 94–98, 100–103, 105–107, 112 Direct By Number ....................................................... 885
file menu ................................................................. 76–79 Number Verification ................................... 866, 868, 870, 873
import/export .................................................................82 NUSN .................................................................................797
initial configuration ......................................................108
offline ...................................................................... 94, 95
preferences ...........................................84, 86, 88–90, 92
O
security mode ............................................................. 122 offline ................................................................................... 94
tools create new config ......................................................... 94
busy on held validation ..........................118 open file ........................................................................ 94
connect to ..............................................116 open file set .................................................................. 95
export user ............................................ 116 receive config ............................................................... 95
extension renumber ...............................115 send config ................................................................... 95
import templates ....................................119 on-boarding: configuring SSL VPN ....................................625
license migration ................................... 120 open configuration ............................................................... 76
line renumber ........................................ 115 open file ............................................................................... 94
MSN configuration .................................118 open file set ......................................................................... 95
print button labels .................................. 118 opening a configuration ....................................................... 46
SCN service user management ............ 117 login messages .............................................................48
view .............................................................................114 operator ............................................................................. 202
Menu Loop Count .............................................................. 885 Optional Greeting ...............................................................885
menu programming ............................................................466 Outbound Contact Express
message waiting indication ................................................737 dialer ...........................................................................282
Migrate outgoing call
Subscription ................................................................ 611 call scenarios ..............................................................845
migrating ADI licenses ....................................................... 684 overflow ............................................................................. 488
mobility ...............................................................................468 overview ...............................................................................35
move .................................................................................. 794
moving borders .................................................................... 62
MS Teams line ................................................................... 334 P
VoIP ............................................................................ 337
panes
MSN configuration ..............................................................118
moving the details pane ................................................63
mult-line options .................................................................457
showing or hiding ..........................................................62
multiple call appearance .................................................. 1123
Park & Page .......................................................................898
music on hold
park and page ....................................................................248
alternate source ..........................................................659
PC requirements ..................................................................42
system source ............................................................ 659
personal directory .............................................................. 474
Personal Meet-Me ............................................................. 923
N PIN
Personal Meet-Me ...................................................... 923
Name PLDS licensing .................................................................. 670
Auto-Attendant ............................................................885 Ports
Match Order ................................................................885 Subscription ................................................................610
Recording name prompts ........................................... 908 Pre-recorded prompt files .................................................. 907

June 2024 Administering Avaya IP Office using Manager 1251

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preferences ..........................................................................84 Replay Menu ......................................................................900
directories ..................................................................... 86 Request Methods
discovery ...................................................................... 88 SIP ..............................................................................877
preferences ...................................................................84 Reseller ............................................................................1242
security ......................................................................... 90 resilience ............................................................................806
validation ...................................................................... 92 resizing the manager window .............................................. 61
visual preferences ........................................................ 89 Response Methods
PRI trunks .......................................................................... 343 SIP ..............................................................................877
E1 line .........................................................................344 Restore .............................................................................. 607
E1 line channels ......................................................... 350 retrieve ...............................................................................794
E1 R2 Advanced .........................................................356 RFC ................................................................................... 875
E1 R2 channels .......................................................... 354 ring tones ........................................................................... 253
E1 R2 line ................................................................... 352 Ringback tone ....................................................................862
E1 R2 MFC group .......................................................356 RIP dynamic routing .......................................................... 545
T1 channels ................................................................ 360
T1 line .........................................................................358
T1 PRI line ..................................................................363
S
print button labels ...............................................................118 S0 line ................................................................................408
Privacy ............................................................................... 716 sales ................................................................................ 1243
Consent ...................................................................... 883 save configuration ................................................................77
Programmable buttons .................................................... 1042 save configuration as ...........................................................78
Prompt SCN service user management ......................................... 117
Name prompts ............................................................ 908 SDKs ................................................................................1243
No Match Prompt ........................................................885 security field .......................................................................141
Pre-recorded ...............................................................907 general ........................................................................142
Short code .................................................................. 906 rights groups
TTS .............................................................................908 configuration ......................................... 161
Prompts group details ......................................... 161
Announcements security administration .......................... 162
Text-to-Speech ......................................881 service users .............................................................. 167
Text-to-Speech ........................................................... 881 services settings ......................................................... 158
system
Q certificates .............................................150
system details ....................................... 147
Quick Reference Guides ..................................................1242 unsecured interfaces .............................149
quota .................................................................................. 511 system status
external ................................................. 166
HTTP .....................................................164
R security administration .......................... 163
reboot ...................................................................................97 Telephony APIs ..................................... 163
receive config .......................................................................95 web services ......................................... 164
Receptionist security mode menus .........................................................122
Subscription ................................................................606 security preferences ............................................................ 90
record consolidation .............................................................70 security settings ................................................................. 102
Recording erase ...........................................................................102
Consent ...................................................................... 883 self-administration ..............................................................476
Recording prompts ............................................................ 906 send config .......................................................................... 95
recordings server edition
user playback ............................................................. 476 default settings ............................................................. 69
recreate SD card ................................................................105 record consolidation ..................................................... 70
Remote access .................................................................. 607 solution view ........................................................... 65, 67
remote access server configuration fields ......................... 519 system inventories ........................................................69
PPP ............................................................................ 519 user interface ................................................................65
Remote hot desking ...........................................................760 server edition licenses
Remote Operations ............................................................286 distributing .................................................................. 672
remote server .....................................................................555 Server Name Indication ..................................................... 398
Remote Support Services ..................................................506 service ............................................................................... 731

June 2024 Administering Avaya IP Office using Manager 1252

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service configuration fields ........................................ 505, 506 SLIC ...................................................................................398
autoconnect ................................................................ 511 SM line ...............................................................................400
bandwidth ................................................................... 507 T38 Fax ...................................................................... 407
dial in .......................................................................... 515 VoIP ............................................................................ 403
fallback ....................................................................... 514 SMDR ...................................................................... 272, 1206
IP ................................................................................ 510 examples .................................................................. 1214
PPP ............................................................................ 512 field descriptions .......................................................1209
quota ........................................................................... 511 SMTP .................................................................................271
SSL VPN ............................................................ 515, 516 SNI .....................................................................................398
SSL VPN fallback ....................................................... 518 SNMP alarm ...................................................................... 654
SSL VPN NAPT ..........................................................517 SNTP ................................................................................. 663
SSL VPN session ....................................................... 516 SoftConsole
service users ......................................................................127 Subscription ................................................................606
Settings solution view ........................................................................ 66
Auto-Attendant ............................................................885 add ................................................................................69
SHAKEN .................................................... 866, 868, 870, 873 display .......................................................................... 66
Short code Set all nodes to Select ..................................................68
Auto-attendant ............................................................ 910 Set all nodes to Subscription ........................................ 68
Short Code set license source .........................................................68
Auto-attendant prompt ................................................906 source numbers ................................................................. 449
short code configuration fields ........................................... 503 Source Numbers ................................................................795
short code feature Speak By Name .................................................................901
auto intercom deny off ................................................ 957 Recording name prompts ........................................... 908
auto intercom deny on ................................................ 957 Speak By Number ..............................................................902
short codes ........................................................................ 448 Speech AI .................................................................. 882, 885
simplified view .................................................................... 114 Speech Voice .....................................................................885
Simultaneous ..................................................................... 793 SRTP ................................................................................. 137
SIP ............................................................................. 474, 877 SSL VPN ....................................................................515, 516
Headers ...................................................................... 878 SSL VPN fallback .............................................................. 518
Hold Reminders ..........................................................863 SSL VPN NAPT ................................................................. 517
Request Methods ....................................................... 877 SSL VPN session .............................................................. 516
Response Methods .....................................................877 Station Message Detail Reporting ................................... 1206
RFC ............................................................................ 875 examples .................................................................. 1214
Ringback tone .............................................................862 field descriptions .......................................................1209
STIR/SHAKEN ....................................866, 868, 870, 873 status bar ............................................................................. 60
Tag length ................................................................... 878 steal ................................................................................... 794
SIP advanced .................................................................... 392 Steal
SIP credentials ...........................................................342, 391 Button ....................................................................... 1078
SIP DECT base ................................................................. 432 STIR ...........................................................866, 868, 870, 873
SIP engineering ......................................................... 341, 398 subscription
SIP extension VoIP ............................................................ 425 error mode .................................................................. 608
SIP line .............................................................................. 372 expiry .......................................................................... 608
Codec renegotiation ................................................... 398 grace period ................................................................608
Keepalives .................................................................. 398 Subscription
SNI ..............................................................................398 DNS ............................................................................ 609
sip line appearances ..........................................................383 Internet Access ...........................................................609
SIP messaging ...................................................................857 IP Route ......................................................................609
SIP prefix ................................................................... 848, 851 Migrate to .................................................................... 611
SIP REFER ........................................................................861 Ports ........................................................................... 610
SIP T38 Fax .......................................................................390 Time Source ............................................................... 609
SIP transport ......................................................................376 subscription configuration fields .........................................604
SIP trunk subscriptions ......................................................................550
configuring .................................................................. 831 Subscriptions
overview ..................................................................... 831 Applications ................................................................ 606
sip uri ................................................................................. 380 CTI ..............................................................................606
SIP URI ..............................................................................379 Media Manager ...........................................................606
SIP VoIP ............................................................................ 386 Receptionist ................................................................606

June 2024 Administering Avaya IP Office using Manager 1253

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Subscriptions (continued) System Phone ................................................................... 666
SoftConsole ................................................................ 606 system security fields .........................................................146
Telephony User ...........................................................605 certificates .................................................................. 150
Telephony User Plus ...................................................605 rights groups
Trial Mode ...................................................................605 configuration ......................................... 161
Unified Communications User .................................... 605 group details ......................................... 161
User Subscriptions ..................................................... 605 security administration .......................... 162
supervised transfer ............................................................ 782 service users .............................................................. 167
Supervised Transfer ...........................................................897 services settings ......................................................... 158
supervisors settings ........................................................... 453 system details .............................................................147
support .............................................................................1243 system status
Syslog alarm ...................................................................... 654 external ................................................. 166
System Administrator .......................................................1242 HTTP .....................................................164
System Alarm .................................................................... 654 security administration .......................... 163
System Conference Telephony APIs ..................................... 163
Add ............................................................................. 927 web services ......................................... 164
Delete ......................................................................... 928 unsecured interfaces .................................................. 149
Edit ............................................................................. 928 system shutdown ................................................................. 97
Settings .......................................................................929 System Status ....................................................................107
System Conferences ......................................................... 927
system configuration field
Avaya Cloud Services .................................................286
T
Contact Center ........................................................... 284 T1 line ................................................................................ 358
system configuration fields ................................................ 203 channels ..................................................................... 360
Access Control Lists ................................................... 282 T1 PRI line ......................................................................... 363
call log ........................................................................ 255 T38 fax ...............................................................................429
customer call reporter .................................................276 tabs
DHCP pools ................................................................228 changing display of .......................................................63
dialer ...........................................................................282 Tag length .......................................................................... 878
DNS ............................................................................ 230 TCP Tunnels ...................................................................... 506
HTTP directory services ............................................. 263 Technical Bulletins ........................................................... 1243
LAN settings ............................................................... 213 telephony ................................................................... 239, 450
LAN VoIP .................................................................... 215 call log ................................................................ 255, 459
LAN1 ...........................................................................213 call settings .................................................................450
LAN2 ...........................................................................230 mult-line options ......................................................... 457
LDAP directory services ............................................. 259 park and page .............................................................248
network topology ........................................................ 223 ring tones ....................................................................253
ring tones ....................................................................253 supervisor settings ......................................................453
SMDR ......................................................................... 272 System Manager ................................................ 253, 254
SMTP ..........................................................................271 tones and music ......................................................... 249
system ........................................................................ 203 TUI ......................................................................256, 460
system events .............................................................265 Telephony User ..................................................................605
system events alarms .................................................266 Telephony User Plus ..........................................................605
system events configuration ....................................... 265 template
System Manager ................................................ 253, 254 analog trunk ................................................................689
telephony ......................................................239 creating .......................................................................687
park and page ....................................... 248 templates
tones and music ......................................................... 249 creating a new record from a template ....................... 688
TUI ..............................................................................256 Text-to-Speech ...........................................................881, 885
voice compression modules ....................................... 274 Recording a Prompt ....................................................908
voicemail .....................................................................232 TFTP log ............................................................................ 114
VoIP ............................................................................ 277 Time ................................................................................... 663
VoIP security ...............................................................279 Manual ........................................................................666
system events ............................................................265, 636 Subscription ................................................................609
system events alarms ........................................................ 266 System Status ............................................................ 665
system events configuration .............................................. 265 time profile configuration fields .................................. 537, 667
System Manager ........................................................253, 254 title bar ................................................................................. 52

June 2024 Administering Avaya IP Office using Manager 1254

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tones and music .................................................................249 telephony (continued)
toolbars ........................................................................ 52, 114 mobility ....................................................................... 468
details ........................................................................... 54 personal directory ....................................................... 474
main ..............................................................................53 short codes ................................................................. 448
moving .......................................................................... 62 SIP ..............................................................................474
navigation ..................................................................... 54 source numbers ..........................................................449
showing or hiding ..........................................................62 telephony ......................................................450
tools menu ..........................................................................115 call log ...................................................459
tooltip ..................................................................................114 call settings ........................................... 450
training ................................................................... 1243, 1244 mult-line options ....................................457
transfer ...............................................................................782 supervisor settings ................................ 453
transfer return .................................................................... 782 TUI ........................................................ 460
Transfer to Auto Attendant .................................................904 user .............................................................................433
Transport Protocols voice recording ........................................................... 465
SIP ..............................................................................877 voicemail .....................................................................442
Trial Mode User Guides .....................................................................1242
Subscription ................................................................605 user interface ....................................................................... 52
trunk templates .................................................................. 686 configuring
applying a template to an analog trunk .......................689 icon size .................................................. 63
importing .....................................................................687 moving borders ....................................... 62
TTS .................................................................................... 881 moving the details pane .......................... 63
Enable ........................................................................ 882 moving toolbars .......................................62
Recording a prompt .................................................... 908 resizing the manager window ................. 61
Speech AI ................................................................... 885 showing or hiding panes ......................... 62
TUI ............................................................................. 256, 460 showing or hiding toolbars ...................... 62
tunnel configuration fields .................................. 559, 561–564 tab display ...............................................63
twinning ......................................................................468, 724 default settings ............................................................. 69
details pane .................................................................. 57
error pane .......................................................59
U automatic validation settings ...................59
Unified Communications User ........................................... 605 revalidating settings ................................ 60
unsupervised transfer ........................................................ 782 viewing error ........................................... 60
Unsupervised Transfer .......................................................903 group pane ..................................................... 55
upgrade ................................................................................98 add record ...............................................56
changing the .bin file directory ....................................100 columns displayed .................................. 56
searching for systems .................................................100 delete record ........................................... 56
user ....................................................................................433 show in groups ........................................57
button programming ................................................... 476 sorting ..................................................... 55
No User .............................................................. 729, 730 validate record ........................................ 57
self-administration ...................................................... 476 navigation pane ............................................................ 54
subscriptions ...............................................................550 server edition .................................................... 65, 67, 69
suppressing NoCallerId alarm .................................... 730 solution view ........................................................... 65, 67
web self-administration ...............................................476 status bar ......................................................................60
User system inventories ........................................................69
Call Barring .................................................................720 title bar ..........................................................................52
NoUser Source Numbers ........................................... 797 toolbars .........................................................................52
Source Numbers .........................................................795 toolsbars
user configuration fields .....................................................433 details ......................................................54
announcements .......................................................... 472 main ........................................................ 53
button programming ................................................... 466 navigation ................................................54
dial in .......................................................................... 465 user management overview ...............................................713
DND ............................................................................448 user portal ..........................................................................476
forwarding ...................................................................461 user rights configuration fields ........................................... 579
hunt group membership ..............................................472 forwarding ...................................................................588
menu programming ...................................... 466 telephony ......................................................581
4400/6400 ............................................. 467 call log ...................................................585
huntgroup ..............................................467 call settings ........................................... 581

June 2024 Administering Avaya IP Office using Manager 1255

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telephony (continued)
telephony (continued)
multi-line options ................................... 584
supervisor settings ................................ 582
user .............................................................................579
user rights membership ...................................... 586, 587

V
validation preferences ..........................................................92
visual preferences ................................................................89
VM locales ..........................................................................112
voice compression modules .............................................. 274
voice recording .................................. 465, 498, 527, 528, 548
voicemail ............................................................232, 442, 493
Voicemail Pro
Auto-Attendant ............................................................880
Time ............................................................................663
VoIP ........................................................................... 277, 282
VoIP security ......................................................................279

W
WAN port configuration fields ............................................ 530
advanced .................................................................... 533
DLCI ........................................................................... 532
frame relay ..................................................................531
Web License Manager ....................................................... 671
web self-administration ...................................................... 476
WebLM ...............................................................................671
installing a license file .................................................678
WebLM host ID .................................................................. 678
websites ...........................................................................1243

June 2024 Administering Avaya IP Office using Manager 1256

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