Developer and Solution Partner

Program
Inter-Working Report

Partner: ALE China

Solution name: M3, M5, M7
Alcatel-Lucent Enterprise Platform:
OmniPCX Enterprise

January 2021
Legal notice
The Alcatel-Lucent name and logo are trademarks of Nokia used under license by ALE. To view other
trademarks used by affiliated companies of ALE Holding, visit: www.al-enterprise.com/en/legal/
trademarks-copyright. All other trademarks are the property of their respective owners.
The information presented is subject to change without notice. Neither ALE Holding nor any of its
affiliates assumes any responsibility for inaccuracies contained herein.
© 2021 ALE International. All rights reserved. http://www.al-enterprise.com

Disclaimer
The product and release listed have been tested with the Alcatel-Lucent Enterprise Platform and the
release specified hereinafter. The tests concern only the inter-working between the DSPP member’s
product and the Alcatel-Lucent Enterprise Platform referenced above. The inter-working report is valid
until the DSPP member’s product issues a new major release of such product (incorporating new
features or functionality), or until ALE issues a new major release of such Alcatel-Lucent Enterprise
product (incorporating new features or functionalities), whichever first occurs.
While efforts were made to verify the completeness and accuracy of the information contained in this
documentation, this document is provided “as is”.
In the interest of continued product development, ALE International reserves the right to make
improvements to this documentation and the products it describes at any time, without notice or
obligation.

Document history
Revision Date Author Details
1 March 2020 Durgadevi Subash Creation
Karthik Padmarajan
2 Nov 2020 Durgadevi Subash Updated the results for few
testcases in the new firmware
version
3 Dec 2020 Rachid HIMMI Update after internal review
4 Jan 2021 Durgadevi Subash Updated the results for Spatial
and PCS tests.

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Tests Overview

Date March 2020 / November 2020 /January 2021

ALE representative Chevert Thierry
Partner representative Lea LI
ALE platform OmniPCX Enterprise
ALE release R12.3.1 / R12.4
Partner solution Myriad series
Partner release 2.10.13.1118 / 2.11.01.1602
Solution categories Office phone

Tests results
Passed Passed with restriction Postponed Refused

Refer to the section 4 for a summary of the test results.

IWR validity extension

 None

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Contact information

Name Lea LI
Title Marketing manager
Address 2nd Floor, Building 1, No 60, NaXian Road, Pudong
New District
Zip code
City SHANGHAI
Country CHINA
Phone + 86 21 2059 9439
Fax
Mobile phone
Email address [email protected]
Web site www.aledevice.com

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 Table of
contents

1 INTRODUCTION .................................................................... 7
1.1 Definition ............................................................................................................ 7
1.2 Validity of the InterWorking Report .................................................................. 7
1.3 Limit of the technical support ........................................................................... 8
1.3.1 Case of additional Third-party applications ........................................................ 8

2 SOLUTION INFORMATION ................................................... 9

3 TEST ENVIRONMENT ......................................................... 10

3.1 Hardware configuration ................................................................................... 10
3.2 Software configuration .................................................................................... 11

4 SUMMARY OF TESTS ......................................................... 12

4.1 Summary of main functions supported ......................................................... 12
4.2 Summary of problems ..................................................................................... 13
4.3 Summary of limitations ................................................................................... 13
4.4 Notes, remarks ................................................................................................. 14

5 TESTS RESULT ................................................................... 15

5.1 Template ........................................................................................................... 15
5.2 Preliminary tests .............................................................................................. 16
5.2.1 Test Objectives ................................................................................................ 16
5.2.2 Test Results Connectivity Setup ...................................................................... 16
5.3 Duplication and Robustness........................................................................... 18
5.4 Audio codecs negotiations/ VAD / Framing................................................... 26
5.5 Outgoing Calls ................................................................................................. 29
5.6 Incoming Calls ................................................................................................. 34
5.7 Features during Conversation ........................................................................ 43
5.8 Call Transfer ..................................................................................................... 47

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 5.8.1 Unattended transfers ........................................................................................ 48
5.8.2 Semi-Attended Transfer (on Ringing) ............................................................... 49
5.8.3 Attended Transfer (in Conversation) ................................................................ 50
5.9 Attendant .......................................................................................................... 51
5.10 Voice Mail ......................................................................................................... 55

6 Appendix A: SOLUTION DESCRIPTION ........................... 57

7 Appendix B: PARTNER side CONFIGURATION ............... 60

8 Appendix C: ALE side CONFIGURATION ......................... 70

8.1 SIP gateway ...................................................................................................... 70
8.2 SIP Proxy .......................................................................................................... 71
8.3 Codec configuration ........................................................................................ 71
8.4 OXE Domain ..................................................................................................... 72
8.5 SIP User Configuration .................................................................................... 72
8.6 Licensing .......................................................................................................... 72

9 Appendix D: SUPPORT PROCESS .................................... 73

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 Chapter

1 1 INTRODUCTION

1.1 Definition

This document is the result of the certification tests performed between the DSPP member’s solution and
Alcatel-Lucent Enterprise’s platform.

It certifies proper inter-working with the DSPP member’s solution.

Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, ALE cannot guarantee accuracy of printed
material after the date of certification nor can it accept responsibility for errors or omissions. Updates to this
document can be viewed on:

- the Technical Support page of the Enterprise Business Portal (https://businessportal.alcatel-lucent.com)

in the Interworking Reports corner (access is restricted to Business Partners and DSPP members )

1.2 Validity of the InterWorking Report

This InterWorking report specifies the products and releases which have been certified.

This inter-working report is valid unless specified until the DSPP member issues a new major release of such
product (incorporating new features or functionalities), or until ALE issues a new major release of such Alcatel-
Lucent Enterprise product (incorporating new features or functionalities), whichever first occurs.

A new release is identified as following:

 a “Major Release” is any x. enumerated release. Example Product 1.0 is a major product release.
 a “Minor Release” is any x.y enumerated release. Example Product 1.1 is a minor product release

The validity of the InterWorking report can be extended to upper major releases, if for example the interface
didn’t evolve, or to other products of the same family range. Please refer to the “IWR validity extension” chapter
at the beginning of the report.

Note 1: The InterWorking report becomes automatically obsolete when the mentioned product releases are end
of life.
Note 2: The renewal of the interoperability test (certification) is under the responsibility of the partner
Note 3: ALE usually generate a major release every 18 or 24 months. Therefore, the IWR is implicitly valid for
two year after the publication.
Chapter 1 1 INTRODUCTION

1.3 Limit of the technical support

For certified DSPP solutions, Technical support will be provided within the scope of the features which have
been certified in the InterWorking report. The scope is defined by the InterWorking report via the tests cases
which have been performed, the conditions and the perimeter of the testing and identified limitations. All those
details are documented in the IWR. The Business Partner must verify an InterWorking Report (see above
“Validity of the InterWorking Report) is valid and that the deployment follows all recommendations and
prerequisites described in the InterWorking Report.

The certification does not verify the functional achievement of the DSPP member’s solution as well as it does not
cover load capacity checks, race conditions and generally speaking any real customer's site conditions.

Access to technical support by the ALE Business Partner requires a valid ALE maintenance contract
rd
For details on all cases (3 party application certified or not, request outside the scope of this IWR, etc.), please
refer to Appendix “DSPP Escalation Process”.

1.3.1 Case of additional Third-party applications

In case at a customer site an additional third-party application NOT provided by ALE is included in the solution
between the certified Alcatel-Lucent Enterprise and DSPP member products such as a Session Border
Controller or a firewall for example, ALE will consider that situation as to that where no IWR exists. ALE will
handle this situation accordingly (for more details, please refer to Appendix “DSPP Escalation Process”).

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 Chapter
Chapter
2 2 SOLUTION INFORMATION
2

Solution name M3/M5/M7

Solution version 2.10.13.1118 / 2.11.01.1602
Interface/API SIP
Interface/API version if relevant

Brief Solution description:

Myriad Series DeskPhone (M3/M5/M7) are SIP-based products that can be integrated into a variety
of solutions with easy provisioning tools in a cost effective, secure and flexible manner
 Chapter

3 3 TEST ENVIRONMENT

Figure 1 Test environment

3.1 Hardware configuration

List main hardware equipments used for testing

 OmniPCX Entreprise:
Chapter 3 1 TEST ENVIRONMENT

o CS (Virtual ESXI server Call Server)

o GD3 (Gateway driver processing Unit)
o PRA T2 (ISDN Access)

OXE setup Node 1

10.9.223.187 &
OXE IP address
10.1.6.1 /10.10.10.11
Domain name etesting2.etesting.lab
Attendant No 9
OXE Extension Details used for test
IP touch and UA extensions 8110 to 8114
SIP users 12023- 12026 & 2110 -2116

3.2 Software configuration

List main softwares used for testing

• Alcatel-Lucent Enterprise Communication Platform: OmniPCX Enterprise R12.3.1 / R12.4

• Partner Application : M3 DeskPhone  2.10.13.1118 & 2.11.01.1602
M5 DeskPhone  2.10.13.1118 & 2.11.01.1602
M7 DeskPhone 2.10.13.1118 & 2.11.01.1602

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 Chapter

4 4 SUMMARY OF TESTS

4.1 Summary of main functions supported

This section is a summary of the main features tested.

Feature Status Comments

IP network Connectivity

SIP Registration

Spatial redundancy with alternate proxy method This section is tested in the firmware
version 2.11.01
Spatial redundancy with alternate DNS method This section is tested in the firmware
version 2.11.01
Switchover to PCS This section is tested in the firmware
version 2.11.01
Codecs Support (G711, G729)

Local/Network calls
This section is tested in the firmware
Call to a forwarded user version 2.11.01

OXE features (call back, voice mail deposit) This section is tested in the firmware
version 2.11.01
Local/Network calls
This section is tested in the firmware
Forward (Immediate / On no reply / On busy) version 2.11.01

OXE features (call pick-up) This section is tested in the firmware

version 2.11.01
Hold / Resume This section is tested in the firmware
version 2.11.01
DTMF sending This section is tested in the firmware
version 2.11.01
Conference This section is tested in the firmware
version 2.11.01.
Unattended transfers

Semi-Attended Transfer (on ringing)

Chapter 4 1 SUMMARY OF TESTS

Attended Transfer (in conversation)

Call to/from an attendant station

Call transfer from an attendant station

Message Waiting Indicator This section is tested in the firmware

version 2.11.01
Forward to voice mail This section is tested in the firmware
version 2.11.01.
Transfer Key in Ringing Conversation

Conference key This section is tested in the firmware

version 2.11.01.
Do Not Disturb Key

4.2 Summary of problems

 NA

4.3 Summary of limitations

 The tests only apply in business mode so other mode like Hotel are not supported.

 No provision to configure alternate SIP Proxy server option in SIP Phone Webpage.

 No Sip digest authentication.

 The terminals are not handled by OmniVista 8770.

For this session of interworking tests, each phone has been manually deployed and configured
using the phone’s web-based management.
The SW of each phone has been also manually upgraded using the phone’s web-management.
Please refer to the dedicated web site to check availability of other deployment, configuration
and maintenance methods and tools: https://www.aledevice.com/site/download

 OXE does not support the following services on SIP devices such as (with suffix, out of a
conversation phase, ...):
o Leave a callback (if called party isn’t a SIP set)
o Perform a loudspeaker announcement (if called isn’t a SIP set)
o Leave a voice message
o Override a forward
o Camp-on
o Contact Center fonctions (CCD agent logon/log off, etc…)
More details on OXE restrictions in the Features List.

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Chapter 4 1 SUMMARY OF TESTS

4.4 Notes, remarks

 M3, M5 and M7 share the same SIP stack so we considered we can reduce the tests suite
without any impact on the quality. Consequently, M5 and M7 have been fully tested whereas M3
has been partially tested.

 The Section 5.6 incoming calls and 5.10 Voice mail (DTMF) related Testcases were executed
in the firmware version 2.11.0.1.1601 and the OXE Version R12.4. The Remaining Testcases
were executed in the firmware version 2.10.13.

 For Spatial and PCS tests, Uploaded the config .xml file in the Myriad Web interface.

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 Chapter

5 5 TESTS RESULT

5.1 Template

The results are presented as indicated in the example below:

Test
Case Test Case N/A OK NOK Comment
Id
Test case 1
1  Action
 Expected result

Test case 2
The application waits
 Action
2 for PBX timer or
 Expected result
phone set hangs up
Test case 3
Relevant only if the
 Action
3 CTI interface is a
 Expected result
direct CSTA link
Test case 4
 Action No indication, no error
4
 Expected result message

… …

Test Case Id: a feature testing may comprise multiple steps depending on its complexity. Each step has to be
completed successfully in order to conform to the test.
Test Case: describes the test case with the detail of the main steps to be executed the and the expected result
N/A: when checked, means the test case is not applicable in the scope of the application
OK: when checked, means the test case performs as expected
NOK: when checked, means the test case has failed. In that case, describe in the field “Comment” the reason for
the failure and the reference number of the issue either on ALE side or on partner side
Comment: to be filled in with any relevant comment. Mandatory in case a test has failed especially the reference
number of the issue.
Chapter 5 1 TESTS RESULT

5.2 Preliminary tests

5.2.1 Test Objectives

The calls are generated to several users belonging to the same network.
Called party can be in different states: free, busy, out of service, do not disturb, etc.
Calls to data devices are refused.
Points to be checked: ringback tones, busy tones, voice during the conversation, display (on caller and called
party), hang-up phase.

Note: dialing will be based on direct dialing number but also using programming numbers on the SIP phone.

5.2.2 Test Results Connectivity Setup

Test
Case Test Case N/A OK NOK Comment
Id
IP network connectivity
1

SIP set network setup with a static IP address

Configure the phone SIPset-1 with a static IP address
A
Check the network connectivity by pinging the phone
and display.
SIP set network setup with a dynamic IP address
Configure the phone SIPset-1 with a dynamic IP
address (given by a DHCP server)
B
Check the network connectivity by pinging the phone
and display.
SIP Registration
2

SIP registration, using OXE MAIN IP address(es)

(without authentication)
A The phone SIPset-1 is configured to register with the
node1 primary main IP address.
Check the phone registration and display.

SIP registration, using OXE as DNS server

(without authentication)

SIPset-1 DNS servers are configured with node1

primary main IP address as primary DNS server and
B
with node1 secondary main IP address as secondary
DNS server.
SIPset-1 is configured to register with the node1
hostname.
Check the phone registration and display.

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Chapter 5 1 TESTS RESULT

SIP registration, using an External DNS server

(without authentication)

SIPset-1 DNS server is configured with the external

C IP address as primary DNS server There is no other
DNS server address configured in SIPset-1. And SIP
SIPset-1 is configured to register with the node1
hostname.
Check the phone registration and display.
Support of “423 Interval Too Brief” (1)
The phone SIPset-1 is configured with a value lower
D than OXE SIP Min Expiration Date.
Check the phone registration and display.

SIP registration with authentication

For this test, register on a Node with authentication
enable (2)
Configure the phone NwkSIPset-1 with node2 main IP
address as SIP registrar and NwkSIPset-1 as
E authentication password.
Check the phone registration and display.

After make the same actions with a wrong password

and check that the phone is rejected.

UDP/TCP signaling
3

Signalling TCP.
Configure your SIP set to use the protocol SIP over
A TCP

Check the registration, and basic calls.

Signalling UDP.
Configure your SIP set to use the protocol SIP over
UDP
B
Check the registration, and basic calls.

Note: all further tests to be made with UDP

Time synchronization
4

NTP registration (if applicable)

The SIP phone SIPset-1 is configured to retrieve the
date and time from the node1 primary main IP
A address.
Check that SIPset-1 retrieves the right date and time
information and displays it.

Keep alive
5

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Chapter 5 1 TESTS RESULT

Keep alive with SIP OPTIONS messages

Configure SIPset-1 to send SIP OPTIONS messages
to OXE (3).
Check that SIPset-1 stay “In service” (with “csipsets”
OXE command) after SIPset-1 sends a SIP
OPTIONS message and receives a response back
A from OXE.
Disconnect SIPset-1 by removing the cable for a
wired phone or the battery for a wireless phone.
Check that SIPset-1 fails to “Out of service” state after
a SIP OPTIONS period (by default 30 seconds).

Notes:

(1) On the SIP client, specify a default registration period inferior to that of OXE SIP registrar (configured via
mgr under SIP/SIP Registrar/SIP Min Expiration Date). OXE will reject with error “423 Interval Too Brief”.
Check that SIP set increases registration period accordingly and the registration happens successfully.

(2) The SIP authentication is configured via mgr under : SIP/SIP Proxy/Minimal authentication method=”SIP
None” or ”SIP Digest”
The SIP keep-alive should be activated on OXE. The configuration is done with “Keep alive” parameter of the
SIP phone classes of service. The SIP keep-alive period is configured in the IP Quality of Service COS.

5.3 Duplication and Robustness.

Check how the system will react in case of a CPU reboot, switchover or link failure etc.
The test system is configured with spatial redundancy (duplicate call servers on two different IP subnetworks).

Spatial redundancy can be configured in two ways:

 “Alternate Proxy method”: Specify both CS MAIN addresses as primary and alternative proxy
respectively. Requires that on non-availability of primary proxy, secondary proxy is used. Requires ability
to accept incoming calls from secondary proxy.
 “DNS method”: Do not specify a proxy address, only SIP domain. Specify the CS MAIN address as first
and second DNS server, respectively. Requires that (at least on non-availability of current proxy) a new
DNS request is issued for every message. Only MAIN CS will respond. Requires ability to accept
incoming calls from secondary CS when it becomes new MAIN.

For each configuration, check:

Can new outgoing calls be made immediately after switchover?
Are existing calls maintained after switchover?
Are incoming calls (from new MAIN CS) accepted immediately after switchover?
Can existing call be modified (transfer, hang-up, etc.) after switchover?
Check if a session that has been started before switchover is maintained after switchover, i.e. does the new
MAIN CS send session updates and is this accepted by the client?

Test
Case Test Case N/A OK NOK Comment
Id

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id

1 Spatial redundancy with alternate proxy method

Spatial redundancy, using “Alternate Proxy

method”, two SIP sets in conversation
Configure SIPset-1, SIPset-2 and SIPset-3 to use two
SIP proxies (OXE call server a and OXE call server b
IP addresses). Configure SIPset-1, SIPset-2 and
SIPset-3 to send SIP OPTIONS keep alive messages
to both SIP proxies.

With SIPset-1 call SIPset-2. Answer the call and check

audio and display.
Configured “Main call
server IP address in
Switchover to standby call server using OXE “bascul”
SIP server and
A command (check first the database replication using
backup call server IP
OXE “twin” command). Check that the existing call is
in Secondary SIP
maintained.
server.
Wait for a “session timer” expiration (this timer is
negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.

With SIPset-1 call SIPset-3. Answer the call and check

that SIPset-2 is put on hold.
With SIPset-1 transfer SIPset-3 to SIPset-2. Check that
the transfer is correctly performed.
Spatial redundancy, using “Alternate Proxy
method”, one SIP set in conversation with a
external party
Configure SIPset-1 and SIPset-2 to use two SIP
proxies (OXE call server a and OXE call server b IP
addresses). Configure SIPset-1 and SIPset-2 to send
SIP OPTIONS keep alive messages to both SIP
proxies.

SIPset-1 call 72 SIPset-2 ( “prefix to take T2

loopback”+target MCDU number>).
B .
Answer the call and check audio and display.
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command). Check that the call is
maintained.

Wait for a “session timer” expiration (this timer is

negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Spatial redundancy, using “Alternate Proxy
method”, call after the switchover
Configure SIPset-1 and SIPset-2 to use two SIP
proxies (OXE call server a and OXE call server b IP
addresses). Configure SIPset-1 and SIPset-2 to send
SIP OPTIONS keep alive messages to both SIP
proxies.
C
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command).

Just after the switchover, with SIPset-1 call SIPset-2.

Answer the call and check audio and display.

Spatial redundancy, using “Alternate Proxy

method”, call after the switchover and a
registration timeout
Configure SIPset-1 and SIPset-2 to use two SIP
proxies (OXE call server a and OXE call server b IP
addresses). Configure SIPset-1 and SIPset-2 to send
SIP OPTIONS keep alive messages to both SIP
proxies.
D
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command).

Wait for a Registration period timeout. After this period,

with SIPset-1 call SIPset-2. Answer the call and check
audio and display.

2 Spatial redundancy with alternate DNS method

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Spatial redundancy, using “DNS method” and OXE
used as DNS server, two SIP sets in conversation
Configure SIPset-1, SIPset-2 and SIPset-3 to use one
SIP proxies (OXE call server node name). Configure
SIPset-1, SIPset-2 and SIPset-3 to use two DNS
servers (OXE call server a and OXE call server b IP
addresses).

With SIPset-1 call SIPset-2. Answer the call and check

audio and display.

Switchover to standby call server using OXE “bascul”

command (check first the database replication using
A
OXE “twin” command). Check that the existing call is
maintained.

Wait for a “session timer” expiration (this timer is

negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.

With SIPset-1 call SIPset-3. Answer the call and check

that SIPset-2 is put on hold.

With SIPset-1 transfer SIPset-3 to SIPset-2. Check that

the transfer is correctly performed.
Spatial redundancy, using “DNS method” and OXE
used as DNS server, one SIP set in conversation
with a external party
Configure SIPset-1 and SIPset-2 to use one SIP
proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use two DNS servers (OXE
call server a and OXE call server b IP addresses).

SIPset-1 call 72 SIPset-2 ( “prefix to take T2

B loopback”+target MCDU number>).
Answer the call and check audio and display.
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command). Check that the call is
maintained.
Wait for a “session timer” expiration (this timer is
negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Spatial redundancy, using “DNS method” and OXE
used as DNS server, call after the switchover
Configure SIPset-1 and SIPset-2 to use one SIP
proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use two DNS servers (OXE
call server a and OXE call server b IP addresses).
C
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command).

Just after the switchover, with SIPset-1 call SIPset-2.

Answer the call and check audio and display.
Spatial redundancy, using “DNS method” and OXE
used as DNS server, call after the switchover and a
registration timeout
Configure SIPset-1 and SIPset-2 to use one SIP
proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use two DNS servers (OXE
call server a and OXE call server b IP addresses).
D
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command).

Wait for a Registration period timeout. After this period,

with SIPset-1 call SIPset-2. Answer the call and check
audio and display.
Spatial redundancy, using “DNS method” with a
delegation DNS server, two SIP sets in
conversation
Configure SIPset-1, SIPset-2 and SIPset-3 to use one
SIP proxies (OXE call server node name). Configure
SIPset-1, SIPset-2 and SIPset-3 to use one DNS
server (An external DNS server delegates the DNS
request to OXE call server a and OXE call server b).

With SIPset-1 call SIPset-2. Answer the call and check

audio and display.

E Switchover to standby call server using OXE “bascul”

command (check first the database replication using
OXE “twin” command). Check that the existing call is
maintained.

Wait for a “session timer” expiration (this timer is

negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.
With SIPset-1 call SIPset-3. Answer the call and check
that SIPset-2 is put on hold.
With SIPset-1 transfer SIPset-3 to SIPset-2. Check that
the transfer is correctly performed.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Spatial redundancy, using “DNS method” with a
delegation DNS server, one SIP set in conversation
with a external party

Configure SIPset-1 and SIPset-2 to use one SIP

proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use one DNS server (An
external DNS server delegates the DNS request to
OXE call server a and OXE call server b).
SIPset-1 call 72 SIPset-2 ( “prefix to take T2
F
loopback”+ target MCDU number>).
Answer the call and check audio and display.
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command). Check that the call is
maintained.
Wait for a “session timer” expiration (this timer is
negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.
Spatial redundancy, using “DNS method” with a
delegation DNS server, call after the switchover

Configure SIPset-1 and SIPset-2 to use one SIP

proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use one DNS server (An
G external DNS server delegates the DNS request to
OXE call server a and OXE call server b).
Switchover to standby call server using OXE “bascul”
command (check first the database replication using
OXE “twin” command).
Just after the switchover, with SIPset-1 call SIPset-2.
Answer the call and check audio and display.
Spatial redundancy, using “DNS method” with a
delegation DNS server, call after the switchover
and a registration timeout

Configure SIPset-1 and SIPset-2 to use one SIP

proxies (OXE call server node name). Configure
SIPset-1 and SIPset-2 to use one DNS server (An
external DNS server delegates the DNS request to
H
OXE call server a and OXE call server b).

Switchover to standby call server using OXE “bascul”

command (check first the database replication using
OXE “twin” command).
Wait for a Registration period timeout. After this period,
with SIPset-1 call SIPset-2. Answer the call and check
audio and display.

3 Switchover to PCS

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Passive call server backup, using “Alternate Proxy
method”, two SIP sets in conversation
Configure SIPset-1, SIPset-2 and SIPset-3 to use two
SIP proxies (OXE call server a and OXE call server
PCS IP addresses). SIPset-1, SIPset-2 and SIPset-3
are part of IP domain with a PCS backup. Configure
SIPset-1, SIPset-2 and SIPset-3 to send SIP
Configure spatial
OPTIONS keep alive messages to both SIP proxies.
setup « Host Name »
With SIPset-1 call SIPset-2. Answer the call and check
in SIP server Field
audio and display.
A and in PCS IP
Stop OXE call server a and call server b. Check that
address. In
the existing call is maintained.
Secondary SIP server
Wait for a “session timer” expiration (this timer is
option.
negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.
With SIPset-1 call SIPset-3. Answer the call and check
that SIPset-2 is put on hold.
With SIPset-1 transfer SIPset-3 to SIPset-2. Check that
the transfer is correctly performed.
Passive call server backup, using “Alternate Proxy
method”, one SIP set in conversation with a
external party
Configure SIPset-1 and SIPset-2 to use two SIP
proxies (OXE call server a and OXE call server PCS IP
addresses). SIPset-1 and SIPset-2 are part of IP
domain with a PCS backup. Configure SIPset-1 and
SIPset-2 to send SIP OPTIONS keep alive messages
to both SIP proxies.
See previous
B
SIPset-1 call 72SIPset-2( “prefix to take T2 comment
loopback”+target MCDU number>).
Answer the call and check audio and display.
Stop OXE call server a and call server b. Check that
the call is maintained.

Wait for a “session timer” expiration (this timer is

negotiated between the INVITE and the OK message).
Check that the call is maintained after this timer
expiration.
Passive call server backup, using “Alternate Proxy
method”, call after the switchover
Configure SIPset-1 and SIPset-2 to use two SIP
proxies (OXE call server a and OXE call server PCS IP
addresses). SIPset-1 and SIPset-2 are part of IP
domain with a PCS backup. Configure SIPset-1 and
See previous
C SIPset-2 to send SIP OPTIONS keep alive messages
comment.
to both SIP proxies.

Stop OXE call server a and call server b.

Just after the switchover, with SIPset-1 call SIPset-2.

Answer the call and check audio and display.

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Test
Case Test Case N/A OK NOK Comment
Id

4 Partner SIP endpoint reboot

Partner SIP set reboot

Reboot SIPset-1. When SIPset-1 comes back in
service, call SIPset-2.
A
Check that SIPset-1 is registered and the call
establishment.

5 Network failure

Temporary Link between OXE and the partner SIP

set
Disconnect the link between SIPset-1 and OXE.
Check that SIPset-1 becomes out of service after a
A keep-alive or a registration period.
When SIPset-1 comes back in service, call SIPset-1.
Reconnect the link between SIPset-1 and OXE.
Check that SIPset-1 becomes in service registration
period.

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5.4 Audio codecs negotiations/ VAD / Framing

These tests check that the phones are using the configured and negotiated audio parameters (codec, VAD,
framing).

Base Configuration :
Phones:
Configure the phones to use G.711 A-law, G.711 µ-law, G.729, G.723.1 in this order.
Configure the phones to use framing=20ms (G.711 and G.729) and 30ms (G.723.1).
Configure the phones to NOT use VAD.

OXE :
Manage 2 IP domains:
Domain1: intra=no compression extra=compression
Domain2: intra=no compression extra=compression
Assign SIPset-1 and OXEset-1 to domain 1.
Assign SIPset-2 and OXEset-2 to domain 2.
Set system law = A-law (1)
Set system compression type = G.729 (2)

Test
Case Test Case N/A OK NOK Comments
Id
1 G 711 A, G 711 µ support (Uncompressed codec)

Call from SIPset-1 to OXEset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 A-law.
Check audio quality
A
Call from OXEset-1 to SIPset-1 (intra-domain)
Check that the call is established using direct RTP in
G711 A-law.
Check audio quality

Set system law = µ-law

Configure the phone to use G.711 µ-law, G.711 A-
law, G.729, G.723.1 in this order

Call from SIPset-1 to OXEset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 µ-law.
B
Check audio quality

Call from OXEset-1 to SIPset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 µ-law.
Check audio quality

2 G 723.1, G 729 support (Compressed codec)

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comments
Id
Call from SIPset-1 to OXEset-2 (extra-domain)
Check that the call is established using direct RTP in
G729.
Check audio quality
A
Call from OXEset-2 to SIPset-1 (extra-domain)
Check that the call is established using direct RTP in
G729.
Check audio quality

Set system compression type = G.723.1

Call from SIPset-1 to OXEset-2 (extra-domain)
Check that the call is established using direct RTP in
G723.1.
Check audio quality
B
Call from OXEset-2 to SIPset-1 (extra-domain)
Check that the call is established using direct RTP in
G723.1.
Check audio quality

3 Voice Activity Detection

Configure SIPset-1 to use VAD

Configure OXEset-1 NOT to use VAD

Call from SIPset-1 to OXEset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 A-law.
Check audio quality

Call from OXEset-1 to SIPset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 A-law.
A Check audio quality

Configure SIPset-1 to use VAD

Configure OXEset-1 to use VAD

Redo the same tests

Configure SIPset-1 NOT to use VAD

Configure OXEset-1 to use VAD

Redo the same tests

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comments
Id
Configure SIPset-1 to use VAD
Configure OXEset-2 NOT to use VAD

Call from SIPset-1 to OXEset-2 (extra-domain)

Check that the call is established using direct RTP in
G729.
Check audio quality

Call from OXEset-2 to SIPset-1 (extra-domain)

Check that the call is established using direct RTP in
G729.
B Check audio quality

Configure SIPset-1 to use VAD

Configure OXEset-2 to use VAD

Redo the same tests

Configure SIPset-1 NOT to use VAD

Configure OXEset-2 to use VAD

Redo the same tests

4 Packet framing

Configure SIPset-1 to use framing=30ms (G.711)

Call from SIPset-1 to OXEset-1 (intra-domain)

Check that the call is established using direct RTP in
G711 A-law.
Check audio quality
A
Call from OXEset-1 to SIPset-1 (intra-domain)
Check that the call is established using direct RTP in
G711 A-law.
Check audio quality

Configure SIPset-1 to use framing=30ms (G.729)

Call from SIPset-1 to OXEset-2 (extra-domain)

Check that the call is established using direct RTP in
G729.
B Check audio quality

Call from OXEset-2 to SIPset-1 (extra-domain)

Check that the call is established using direct RTP in
G729.
Check audio quality

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5.5 Outgoing Calls

Called party can be in different states: free, busy, out of service, do not disturb, etc.
Points to be checked: tones, voice during the conversation, display (name and extension number on caller and
called party), hang-up phase.
OXE SEPLOS prefixes are mandatory for several tests of this section. For more information refer to the
appendix.
By default, all phones are multiline set with two lines.

Note: dialing will be based on direct dialing number but also using programming numbers on the SIP phone.

Test
Case Test Case N/A OK NOK Comment
Id

1 Local/Network calls

Call to a local user

With SIPset-1 call the OXE phone OXEset-1.

A Check that OXEset-1 is ringing. On SIPset-1 check

the ring back tone. On both sets check display
(name and extension number)
Answer the call and check audio and display.
Call to a local user with overlap dialing
With SIPset-1 call the OXE phone OXEset-1.
but,
Dial a first part of the number: call 33 (first part of
OXEset-1 extension number), wait one second and
dial 015.
B
Check that call is transmitted to the OXEset-1which
is ringing.

When the OXEset-1 is ringing, hang-up

Check release and display
Call to a local user with overlap dialing, timeout
With SIPset-1 call the OXE phone OXEset-1.
but,
Dial a first part of the number: call 33 (first part of Phone return back to
C
OXEset-1 extension number), and never dial the the home page.
end of the number in order to have a timeout.

Check time out and display

Call to another SIP set
With the SIPset-1 call the SIPset-2
D
Check the display and audio during all steps (dialing,
ring back tone, conversation, and release).

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call to a local user with SIP proxy Authentication
Check that NwkSIPset-1 sip set configured with
authentication.
E
With NwkSIPset-1 call NwkSIPset-2.

Answer the call, check audio and display.

Call to external number (via T2 loopback)
(Check ring back tone, called party display)
With SIPset-1 dial 72OXEset-1 (“prefix to take T2
loop”+target MCDU number>)
F
Check that OXEset-1 is ringing.
Answer the call and check audio, display and call
release.
SIP session timer expiration
Check if call is maintained after the session timer
expiration:
If possible, configure the “Session timer” on SIPset-1
to 120 seconds.

With SIPset-1 call OXEset-1.

G
Answer the call on OXEset-1and never hang up,
wait for session timer expiration.

Check that call is maintained.

Configure the “Session timer” on SIPset-1 to the

default value.
Call to wrong number
(SIP: “404 Not Found”)
With the SIPset-1 call a wrong number which is not Call failed message is
H
in the dialing plan. displayed in Phone

Check the ring back tone and display.

Call rejected by call handling
(SIP: “183 Progress/487 Request Terminated”)
Try to enable OXE “do not disturb” feature on
SIPset-1 with a wrong password calling 42 (Do not 487 Request
disturb prefix). After the voice guide, enter 1111 Terminated
I
(<sip set wrong password>).

Wait for error ring back tone from OXE.

Check the call is rejected

Local/Network calls – called party is not
2
available
Call to local user with no answer
With SIPset-1 call the OXE phone OXEset-1.
And never answer the call. It shows as Missed
A
Check time out and display. call.

Note that OXEset-1 don’t have a Voice Mail

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call to busy user
(SIP: “486 Busy Here”)
With SIPset-1 call OXEset-1, answer the call, and
don’t hang up.
With SIPset-2 call OXEset-1, answer the call, and
don’t hang up.

With SIPset-3 call OXEset-1 which is busy

B Check the ring back tone and display.

With OXEset-1 call SIPset-1, answer the call, and

don’t hang up.
With SIPset-2 call SIPset-1, answer the call, and
don’t hang up.

With SIPset-3 call SIPset-1 which is busy

Check the ring back tone and display.

Call to user in “Out of Service” state
(SIP: “480 Temporarily Unavailable”)
Disconnect SIPset-2 and wait for SIP deregister 487 request
C With SIPset-1 call SIPset-2 which is in “Out of terminated message
Service State” is displayed in trace

Check the display and ring back tone.

Call to user in “Do not Disturb” (DND) state
(SIP: “ 183 Session progress”):
Dial “42” (Do not disturb prefix) on the OXEset-1 in
order to enable the DND. Wait for acknowledgement
487 request
from OXE
D terminated message
With the SIPset-1 call the OXEset-1.
is displayed in trace.
Check ring back tone and display.

Redial 42 on OXEset-1 to cancel the DND

3 Call release

Call release during an outgoing call, release

done by the partner SIP set

With SIPset-1 call OXEset-1 and don’t answer the

call.
A
With SIPset-1, release the call during the ringing
period. Check that OXEset-1 plays a release tone
and goes in idle mode after some seconds

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call release during an outgoing call, release
done by the OXE set

With SIPset-1 call OXEset-1 and don’t answer the

call.
B
With OXEset-1, reject the call during the ringing
period. Check that SIPset-1 plays a release tone
and goes in idle mode after some seconds

4 Identity secrecy

Calling Line Identity Restriction (CLIR): Local

call to SIP terminal with CLIR activated.
With SIPset-1 call OXEset-1 by dialing 409OXEset-1
(Secret identity prefix + <target MCDU number>) in
A
order to hide SIPset-1 identity.
Check that OXEset-1 is ringing, answer the call and
check that SIPset-1 identity is hidden.

Calling Line Identity Restriction (CLIR): Local

call to a PSTN line with CLIR activated.
With SIPset-1 call OXEset-1 by dialing
40972OXEset-1OXEset-1 (Secret identity prefix
B +“prefix to take T2 loop”+ <target MCDU number>)
in order to hide SIPset-1 identity.

Check that OXEset-1 is ringing, answer the call and

check that SIPset-1 identity is hidden.

5 Call to a forwarded user

Call to local user, immediate forward (CFU).

(SIP: “302 Moved Temporarily”)(1)

On OXEset-1 dial the 51 (Immediate forward prefix ).

After the voice guide, enter SIPset-2 <target MCDU
number>) to activate the CFU. Wait for
acknowledgement from OXE.
A
With SIPset-1 call the OXEset-1.
Check that SIPset-2 is ringing and the display.
Answer the call check audio and hung up.

Dial 41 (Forward cancellation prefix) on OXEset-1

for forward cancellation.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call to local user, forward on no reply (CFNR). (1)

On OXEset-1 dial the 53 (Forward on no reply prefix)

After the voice guide, enter SIPset-2 (<target MCDU
number>) to activate the CFNR. Wait for
acknowledgement from OXE.

With SIPset-1 call the OXEset-1. Check that

OXEset-1 is ringing but don’t answer the call and
B
wait the time out (about 30 sec).

After time out check that SIPset-2 is ringing and

answer the call.

Check the audio and display.

Dial 41 (Forward cancellation prefix) on OXEset-1

for forward cancellation.
Call to local user, forward on busy (CFB). (1)

On OXEset-1 dial the 54 (Forward on busy prefix).

After the voice guide, enter SIPset-2 (<target MCDU
number>) to activate the CFB. Wait for
acknowledgement from OXE.

With SIPset-2 call OXEset-1 and answer the call.

With SIPset-3 call OXEset-1 and answer the call to
C
make it busy.

With SIPset-1 call OXEset-1.

Check that SIPset-2 is ringing and answer the call.

Check the audio and display.

Dial 41 (Forward cancellation prefix) on OXEset-1

for forward cancellation.

6 OXE features (Call back, voice mail deposit)

Call Back on free set

From SIPset-1 call OXEset-1
Dial “5” (Call Back suffix) while OXEset-1 is ringing
A and release the call.
Activate the call back from OXEset-1.
Check that SIPset-1 is ringing, answer the call and
check audio + display.
Voice mail deposit
From SIPset-1 call OXEset-1
Dial “6” (Voice Mail deposit suffix) while OXEset-1 is
B ringing. Leave a message when connected to the .
voice mail and release the call.
Check the voice message on OXEset-1.

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Chapter 5 1 TESTS RESULT

5.6 Incoming Calls

Calls will be generated using the numbers or the name of the SIP user.
SIP terminal will be called in different states: free, busy, out of service, forward.
The states are to be set by the appropriate system prefixes unless otherwise noted.
Points to be checked: tones, voice during the conversation, display (on caller and called party), hang-up phase.
OXE SEPLOS prefixes are mandatory for several tests of this section. For more information refer to the
appendix E

Test
Case Test Case N/A OK NOK Comment
Id
1 Local/Network calls

Local /network call to free SIP terminal

Local: with OXEset-1 call SIPset-1. Check that
SIPset-1 is ringing and answer the call

Check ring back tone, audio and called party

display.

PSTN: with OXEset-1 call SIPset-1 by dialing

72SIPset-1 (prefix to take the T2 loopback+ <target
MCDU number>). Check that SIPset-1 is ringing and
A
answer the call.

Check ring back tone, audio and called party

display.

Network: with OXEset-1 call NwkSIPset-1. Check

that NwkSIPset-1 is ringing and answer the call.

Check ring back tone, audio and called party

display.
Local/network call to busy SIP terminal
Local: With SIPset-2 call SIPset-1 and answer the
call to make it partially busy, don’t hang up.
With SIPset-3 call SIPset-1 and answer the call to
make it fully busy, don’t hang up.
With OXEset-1 call SIPset-1 which is busy

Check the ring back tone and display

B
Network: With SIPset-1 call NwkSIPset-1 and
answer the call to make it partially busy, don’t hang
up.
With SIPset-2 call NwkSIPset-1 and answer the call
to make it fully busy, don’t hang up.

With OXEset-1 call NwkSIPset-1 which is busy

Check ring back tone, and called party display.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local/network call to unplugged SIP terminal
Local: Unplug the SIPset-1 SIP set and call it with
OXEset-1.

Check the ring back tone and display

C
Network: Unplug the NwkSIPset-1 sip set and call it
with OXEset-1

Check the ring back tone and display

Local/network call to SIP terminal in Do Not
Disturb mode (DND), by local feature if
applicable
Local: Enable DND on SIPset-1 and call it with
OXEset-1. Initially Busy status
Check the ring back tone and display and , Please go on
D
Cancel the DND on SIPset-1. hook message is
displayed
Network: Enable DND on NwkSIPset-1 and call it
with OXEset-1
Check the ring back tone and display
Cancel the DND on NwkSIPset-1.
Local/network call to SIP terminal in Do Not
Disturb mode (DND), by system feature, secret
code sent by DTMF (SEPLOS)
Local: Enable DND on SIPset-1 using 42 prefix (Do
not disturb prefix). After the voice guide, enter 0000
(<sip set password>). Wait for acknowledgement
from OXE.

With OXEset-1 call SIPset-1

Check the ring back tone and display
Cancel the DND on SIPset-1 using 42 (Do not
E
disturb prefix).

Network: Enable DND on NwkSIPset-1 using 42

prefix (Do not disturb prefix). After the voice guide,
enter 1234 (<sip set password>).Wait for
acknowledgement from OXE.

With OXEset-1 call NwkSIPset-1

Check the ring back tone and display
Cancel the DND on NwkSIPset-1 using 42 (Do not
disturb prefix).

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local call to SIP terminal in Do Not Disturb
mode (DND), by system feature, prefix and
secret code sent in the INVITE message
(SEPLOS)
Local: Enable DND on SIPset-1 using 420000 (Do
not disturb prefix + <sip set password>). for
F
acknowledgement from OXE.

With OXEset-1 call SIPset-1

Check the ring back tone and display
Cancel the DND on SIPset-1 using 42 (Do not
disturb prefix).
SIP session timer expiration
Check if call is maintained after the session timer
expiration:
Configure the “Session timer” on OXE to 120
seconds (3).

With OXEset-1 call SIPset-1.

G
Answer the call on SIPset-1and never hang up, wait
for session timer expiration.

Check that call is maintained.

Configure the “Session timer” on OXE to the default

value : 1800 seconds (3).
External call to SIP terminal.
Check that external call back number is shown
correctly:
With OXEset-1 dial 72SIPset-1 ( “prefix to take T2
loop”+target MCDU number>)
H
Check that SIPset-1 is ringing and the external call
number is shown correctly
Answer the call and check audio, display and call
release.

2 Forward (Immediate/On no reply/On busy)

Local/network/SIP call to SIP terminal in

immediate forward (CFU) to local user, by local
feature if applicable
Local: On SIPset-1 enable CFU to OXEset-1 With
SIPset-2 call SIPset-1. Check that OXEset-1 is
ringing.
Answer the call and check audio and display.
A Disable CFU on SIPset-1.
Network: On NwkSIPset-1 enable CFU to
NwkOXEset-2.
With SIPset-1 call NwkSIPset-1. Check that
NwkOXEset-2 is ringing.
Answer the call and check audio and display.

Disable CFU on NwkSIPset-1.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local/network/SIP call to SIP terminal in
immediate forward (CFU) to local user, by
system feature (SEPLOS)
Local: On SIPset-1 enable CFU to OXEset-1 using
51 prefix (Immediate forward prefix). After the voice
guide, enter OXEset-1 (<target MCDU number>).
Wait for acknowledgement from OXE.
With SIPset-2 call SIPset-1. Check that OXEset-1 is
ringing.
Answer the call and check audio and display. After entered the
Prefix and password
Disable CFU on SIPset-1 using 41 (Forward in the Phone, Wrong
B cancellation prefix) prefix. (Failed) status
message is displayed.
Network: On NwkSIPset-1 enable CFU to But using prefix
NwkOXEset-2 using 51 prefix (Immediate forward Forward is activated.
prefix ). After the voice guide, enter NwkOXEset-2
(<target MCDU number>).
Wait for acknowledgement from OXE.
With SIPset-1 call NwkSIPset-1. Check that
NwkOXEset-2 is ringing.
Answer the call and check audio and display.

Disable CFU on NwkSIPset-1 using 41 (Forward

cancellation prefix).
PSTN call to SIP terminal in immediate forward
(CFU) to local user, by local feature if applicable
Local: On SIPset-1 enable CFU to OXEset-1 With
SIPset-2 call 72SIPset-1( “prefix to take T2
C loopback”+target MCDU number>).
Check that OXEset-1 is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-1.

PSTN call to SIP terminal in immediate forward
(CFU) to local user, by system feature (SEPLOS)
Local: On SIPset-1 enable CFU to OXEset-1 using
51 prefix (Immediate forward prefix). After the voice
guide, enter OXEset-1 (<target MCDU number>)
Wait for acknowledgement from OXE.
With SIPset-2 call 72SIPset-1( “prefix to take T2
D loopback”+target MCDU number>).
Check that OXEset-1 is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-1 using 41 (Forward

cancellation prefix) prefix.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local/network/SIP call to SIP terminal in
immediate forward (CFU) to network number, by
local feature if applicable
Local: On SIPset-1 enable CFU to NwkSIPset-1
With SIPset-2 call SIPset-1. Check that NwkSIPset-1
is ringing.
Answer the call and check audio and display.
E
Disable CFU on SIPset-1.

Network: On SIPset-1 enable CFU to NwkSIPset-2.

With NwkSIPset-1 call SIPset-1. Check that
NwkSIPset-2 is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-1.

Local/network/SIP call to SIP terminal in
immediate forward (CFU) to network number, by
system feature (SEPLOS)
Local: On SIPset-1 enable CFU to NwkSIPset-1
using 51 prefix (Immediate forward prefix). After the
voice guide, enter NwkSIPset-1(<target MCDU
number>).
Wait for acknowledgement from OXE.
With SIPset-2 call SIPset-1. Check that NwkSIPset-1
is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-1 using 41 (Forward

F
cancellation prefix).

Network: On SIPset-1 enable CFU to NwkSIPset-2

using 51 prefix (Immediate forward prefix). After the
voice guide, enterNwkSIPset-2 (<target MCDU
number>).
Wait for acknowledgement from OXE.
With NwkSIPset-1 call SIPset-1. Check that
NwkSIPset-2 is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-1 using 41 (Forward

cancellation prefix).

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local/network/SIP call to SIP terminal in
immediate forward (CFU) to a SIP user, by local
feature if applicable
Local: On SIPset-2 enable CFU to SIPset-3 With
SIPset-1 call SIPset-2. Check that SIPset-3 is
ringing.
Answer the call and check audio and display.
G
Disable CFU on SIPset-2.

Network: On SIPset-2 enable CFU to NwkSIPset-1.

With SIPset-1 call NwkSIPset-2. Check that
NwkSIPset-1 is ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-2.

Local/network/SIP call to SIP terminal in
immediate forward (CFU) to a SIP user, by
system feature (SEPLOS)
Local: On SIPset-2 enable CFU to SIPset-3 using 51
prefix (Immediate forward prefix). After the voice
guide, enterSIPset-3 (<target MCDU number>).
Wait for acknowledgement from OXE.
With SIPset-1 call SIPset-2. Check that SIPset-3 is
ringing.
Answer the call and check audio and display.

Disable CFU on SIPset-2 using 41 (Forward

H cancellation prefix).

Network: On NwkSIPset-1 enable CFU to

NwkSIPset-2 using 51 prefix (Immediate forward
prefix). After the voice guide, enter NwkSIPset-2
(<target MCDU number>).
Wait for acknowledgement from OXE.
With SIPset-1 call NwkSIPset-1. Check that
NwkSIPset-2 is ringing.
Answer the call and check audio and display.

Disable CFU on NwkSIPset-1 using 41 (Forward

cancellation prefix)
Local call to SIP terminal in “forward on busy”
(CFB) state, by local feature if applicable
On SIPset-2 enable CFB to OXEset-1
With SIPset-2 call SIPset-3.
With SIPset-2 call SIPset-4 to make it busy.
I
With SIPset-1 call SIPset-2 which is busy.
Check that OXEset-1 is ringing
Answer the call and check audio and display.

Disable CFU on SIPset-2.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Local call to SIP terminal in “forward on busy”
(CFB) state, by system feature (SEPLOS)
On SIPset-2 enable CFB to OXEset-1 using 52
prefix (Forward prefix on busy). After the voice
guide, enter OXEset-1 (<target MCDU number>).
Wait for acknowledgement from OXE.
With SIPset-2 call SIPset-3.
J With SIPset-2 call SIPset-4 to make it busy.

With SIPset-1 call SIPset-2 which is busy.

Check that OXEset-1 is ringing
Answer the call and check audio and display.

Disable CFB on SIPset-2 using 41 (Forward

cancellation prefix).
Local call to SIP terminal in “forward on no
reply” (CFNR), by local feature if applicable

On SIPset-2 enable CFNR to OXEset-1

With SIPset-1 call SIPset-2.
Check that SIPset-2 is ringing and don’t answer the
K
call, wait for time out (about 30 seconds).
After time out expiration check that SIPset-2 stops
ringing and that the call is not displayed anymore.
Check the OXEset-1 is ringing, answer the call and
check audio and display.
Disable CFNR on SIPset-2.
Local call to SIP terminal in “forward on no
reply” (CFNR), by system feature (SEPLOS)

On SIPset-2 enable CFNR to OXEset-1 using 53

prefix (Forward prefix on no reply). After the voice
guide, enter OXEset-1 (<target MCDU number>).
Wait for acknowledgement from OXE.
L
With SIPset-1 call SIPset-2.
Check that SIPset-2 is ringing and don’t answer the
call, wait for time out (about 30 seconds).
After time out expiration the OXEset-1 is ringing,
answer the call and check audio and display.
Disable CFNR on SIPset-2 using 41 (Forward
cancellation prefix).

3 Call release

Call release during an incoming call, release

done by the partner SIP set

With OXEset-1 call SIPset-1 and don’t answer the

A
call.
With SIPset-1, reject the call during the ringing
period. Check that OXEset-1 plays a release tone
and goes in idle mode after some seconds

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call release during an incoming call, release
done by the OXE set

With OXEset-1 call SIPset-1 and don’t answer the No Release tone
B
call. heard on SIP Set.
With OXEset-1, release the call during the ringing
period. Check that SIPset-1 plays a release tone
and goes in idle mode after some seconds.

4 Calling name presentation

Calling Line Identity Restriction (CLIR): Local

call to SIP terminal.
With OXEset-1 call SIPset-1 by dialing 409SIPset-1
(Secret identity prefix + <target MCDU number>) in
A
order to hide OXEset-1 identity.
Check that SIPset-1 is ringing, answer the call .
Check that OXEset-1 identity is hidden on both
ringing and conversation periods.
Display: Call to free SIP terminal from user with a
name containing non-ASCII characters. Check
caller display.
With SIPset-2 call SIPset-1 (extension with a name
B containing non-ASCII characters).
Check that SIPset-1 is ringing and check on its
display the name “SIPset-2 éëêèè” is displayed.
Check that non-ASCII characters (éëêèè) are
correctly displayed.
Display: Call to free SIP terminal from user with a
UTF-8 name containing non-ASCII characters.
Check caller display.
With SIPset-7 call SIPset-1 (extension with a name
C containing UTF-8 characters).
Check that SIPset-1 is ringing and check on its
display the name: “SIPset-7_&@#” is displayed.
Check that UTF-8 characters (&@#) are correctly
displayed.

5 Hunt group

SIP set is part of a sequential hunt group (1). Call

to hunt group. Check call/release.
With SIPset-4 call the sequential hunt group MCDU
number (33211).
Check that SIPset-1 is ringing
Answer the call and don’t hang up.
And with SIPset-5 call the sequential hunt group
A
MCDU number (33211).
Check that SIPset-2 is ringing
Answer the call and hang up.
And with SIPset-6 call the sequential hunt group
MCDU number (33211).
Check that SIPset-3 is ringing
Answer the call and don’t hang up.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
SIP set is part of a cyclic hunt group (2).
Call to hunt group. Check call/release.
With SIPset-1 call the cyclic hunt group MCDU
number (33212).
Check that SIPset-5 is ringing
Answer the call and hang up.

And with SIPset-1 call the cyclic hunt group MCDU

B
number (33212).
Check that SIPset-6 is ringing
Answer the call and don’t hang up.

And with SIPset-1 call the cyclic hunt group MCDU

number (33212).
Check that SIPset-7 is ringing
Answer the call and hang up.

5 Twin set

SIP set is declared as a twin set (tandem).

Call to main set and see if twin set rings. Take call
with twin set.
With SIPset-1 call OXEset-2 which is in tandem with
SIPset-4.
A
Check that OXEset-2 and SIPset-4 are both ringing.

Answer the call from SIPset-4 and check that

OXEset-2 stop ringing.
Check audio and display.

6 OXE features (Camp-on, call pick-up)

Call to busy user, Call waiting. (Camp-on),

system feature (SEPLOS)
With SIPset-1 call OXEset-1, answer the call and
don’t hang up.
With SIPset-2 call OXEset-1 to make it busy, answer
the call and don’t hang up.
A
With SIPset-3 call OXEset-1 and on SIPset-3 use
the camp-on suffix.

Check the tones and display on SIPset-3.

On OXEset-1 answer the call and check audio,
display.
Call Pick-up.
With SIPset-1 call SIPset-2.
While SIPset-2 is ringing, pick-up the call from
B SIPset-3 by dialing the pick-up prefix + SIPset-2

Check audio and display.

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Chapter 5 1 TESTS RESULT

5.7 Features during Conversation

Features during conversation between OXE user and SIP user must be checked.
Check that right tones are generated on the SIP phone. A multiline SIP set is mandatory for tests 2, 3, 4 and 8.
OXE SEPLOS prefixes are mandatory for several tests of this section. For more information refer to the
appendix E.

Test
Case Test Case N/A OK NOK Comment
Id

1 Hold/resume

Hold and resume in case of a single call (by local

feature if applicable)

With SIPset-4 call OXEset-1

Answer the call, check audio and display.
A
With SIPset-4 put OXEset-1 on hold with “Hold” key,
check tones and display on both sets, then press
again “Hold” key to resume the call (applicable if
Hold Key is provided by the SIP set)

On OXEset-1 put SIPset-4 on hold then resume.

Hold and resume in case of a second call

SIPset-4 (which is multi-lines) is in conversation with

an external PSTN user.

From SIPset-4 , call SIPset-1 and answers the call.

Check that External PSTN user is put on Hold +
tones and display.

Check audio and display on SIPset-4 SIPset-1

B
On SIPset-4, toggle between the external PSTN
user line and SIPset-1 line (via line key)

Check hold tone, audio and display on the sets.

Release the call from SIPset-4 and switch to the first

line.
Check that SIPset-1 and SIPset-4 are in
conversation + display

2 Call release

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Call release during conversation, release done
by the partner SIP set

With SIPset-1 call OXEset-1

A
Answer the call, check audio and display.
With SIPset-1, release the call. Check that OXEset-1
plays a release tone and goes in idle mode after
some seconds
Call release during conversation, release done
by the OXE set

With SIPset-1 call OXEset-1

B
Answer the call, check audio and display.
With OXEset-1, release the call. Check that SIPset-1
plays a release tone and goes in idle mode after
some seconds

3 Call park

Call park with SEPLOS feature

With SIPset-1 call OXEset-1 and answer the call.

On the SIP set SIPset-1 park the call with OXEset-1
by dialing the 402 (call park prefix). After the voice
guide, enter OXEset-2 (<target MCDU number>)
Check that OXEset-1 is put on hold.
Release SIPset-1 but not OXEset-1. Check that
A
OXEset-1 is still on hold.
From SIPset-1, call SIPset-2 answer the call, check
audio and release it.
From SIPset-2 dial 402 OXEset-2 (call park prefix).
After the voice guide, enter OXEset-2 (<OXEset-2
number>).
Check that SIPset-2 and OXEset-1 are in
conversation + display.

4 DTMF sending

Sending DTMF

Configure SIPset-1 to send DTMF using RFC 2833

A From SIPset-1 call the node 1 voice mail directory
number and try to navigate in its menu listed by the
voice guide.
Check that you can navigate in the menus.

5 Conference

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Meet Me conference

With SIPset-1 call 509 (Meet me conference prefix)

After the voice guide, enter SIPset-1 (<target
extension>). After the voice guide, enter 0000
(conference password), don’t release this call.
With OXEset-1 call 509 (Meet me conference prefix)
After the voice guide, enter SIPset-1 (<target
extension>). After the voice guide, enter 0000
(conference password), don’t release this call.
Not tested due to
A Check that OXEset-1 and SIPset-1 are in
setup unavailability.
conference.
With SIPset-1 call 509 (Meet me conference prefix)
After the voice guide, enter SIPset-1 (<target
extension>). After the voice guide, enter 0000
(conference password), don’t release this call.
Check that OXEset-1, SIPset-1 and SIPset-2 are in
conference.
Check audio and display.
Release the conference from OXEset-1. Check that
SIPset-1 and SIPset-2 are in conference.
Three party conference initiated from OXE set

With OXEset-1 call SIPset-1, answer the call and

don’t release it.
With OXEset-1 call SIPset-2, answer the call and
don’t release it too.
B With OXEset-1, the conference bridge, start a
conference.
Check that OXEset-1, SIPset-1 and SIPset-2 are in
conference.
Check audio and display.
Release the conference from OXEset-1. Check that
SIPset-1 and SIPset-2 are in conversation.

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Three party conference initiated from SIP set
(applicable to a SIP multi-line set only)

With SIPset-1 call an external PSTN user, answer

the call and don’t release it.
With SIPset-1 call SIPset-2, answer the call and
don’t release it too.
With SIPset-1, the conference bridge, start a
conference by the local feature.
Check that an external PSTN user, SIPset-1 and
SIPset-2 are in conference.
Check audio and display.
Not tested due to
C Release the conference from SIPset-1. Check that
setup unavailability
the external PSTN user and SIPset-2 are in
conversation.
With SIPset-2 call SIPset-1, answer the call and
don’t release it.
With SIPset-2, the conference bridge, start a
conference by the local feature.
Check that an external PSTN user, SIPset-1 and
SIPset-2 are in conference.
Check audio and display.
Release the conference from SIPset-1. Check that
the external PSTN user and SIPset-2 are in
conversation.

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Chapter 5 1 TESTS RESULT

5.8 Call Transfer

During the consultation call step, the transfer service can be requested and must be tested.
Several transfer services exist: Unattended Transfer, Semi-Attended Transfer and Attended Transfer.
Audio, tones and display must be checked.

We use the following scenario, terminology and notation:

There are three actors in a given transfer event:

 A – Transferee: the party being transferred to the Transfer Target.
 B – Transferor: the party doing the transfer.
 C – Transfer Target: the new party being introduced into a call with the Transferee.

There are three kinds of transfers in the SIP world:

 Unattended Transfer or Basic Transfer: The Transferor provides the Transfer Target's contact to the
Transferee. The Transferee attempts to establish a session using that contact and reports the results of
that attempt to the Transferor.
Note: Unattended Transfer is not provided by the OXE, but might by supported by the SIP set.

 Semi-Attended Transfer or Early Attended Transfer or Transfer on ringing:

1. A (Transferee) calls B (Transferor). A and B in conversation.
2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. C is in ringing state (does not
pick up the call).
3. B executes the transfer. B drops out of the communication. A is now in contact with C, in ringing
state. When C picks up the call it is in conversation with A.

 Attended Transfer or Consultative Transfer or Transfer in conversation:

1. A (Transferee) calls B (Transferor). A and B in conversation.
2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. C picks up the call and goes
in conversation with B.
3. B executes the transfer. B drops out of the communication. A is now in conversation with C.

In the below tables, SIP means a partner SIP set, OXE means a proprietary OXE (Z/UA/IP) set, Ext. Call means
an External Call, PSTN for example

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5.8.1 Unattended transfers

If unattended transfers are available when the transferor is the partner SIP set, describe here the procedure to
initiate such transfer.

 A – Transferee: the party being transferred to the Transfer Target.

 B – Transferor: the party doing the transfer.

Test Test Case N/A OK NOK Comment

Case
Id
A B C
Transferee Transferor Transfer Target
Type of MCDU Type MCDU Type of MCDU
set (2) Number of set Number set (2) Number
(2)
OXE/Ext. OXE/Ext. OXEset-
1 OXEset-1 SIP SIPset-1
Call Call 3
Unavailable
OXE/Ext. OXEset-
2 SIP SIPset-1 OXE OXEset-1 from an OXE
Call 3
set
OXE/Ext. OXEset-1 OXE OXEset-3 SIP SIPset- Unavailable
3 Call 1 from an OXE
set
SIP SIPset-1 OXE OXEset-1 SIP SIPset- Unavailable
4 2 from an OXE
set
OXE/Ext.
5 OXEset-1 SIP SIPset-1 SIP SIPset-2
Call
OXE/Ext. OXEset-
6 SIP SIPset-1 SIP SIPset-2
Call 1
7 SIP SIPset-1 SIP SIPset-2 SIP SIPset-3
 C – Transfer Target: the new party being introduced into a call with the Transferee.

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5.8.2 Semi-Attended Transfer (on Ringing)

If semi attended transfers are available when the transferor is the partner SIP set, describe here the procedure
to initiate such transfer.

Test
Case Test Case N/A OK NOK Comment
Id
A B C
Transferee Transferor Transfer Target
Type of MCDU Type MCDU Type of MCDU
set (2) Number of set Number set (2) Number
(2)
OXE/Ext. OXE/Ext. OXEset-
1 OXEset-1 SIP SIPset-1
Call Call 3
OXE/Ext. OXEset-
2 SIP SIPset-1 OXE OXEset-1
Call 3
OXE/Ext. OXEset-1 OXE OXEset-3 SIP SIPset-
3
Call 1
SIP SIPset-1 OXE OXEset-1 SIP SIPset-
4
2
OXE/Ext.
5 OXEset-1 SIP SIPset-1 SIP SIPset-2
Call
OXE/Ext. OXEset-
6 SIP SIPset-1 SIP SIPset-2
Call 1
7 SIP SIPset-1 SIP SIPset-2 SIP SIPset-3

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5.8.3 Attended Transfer (in Conversation)

If attended transfers are available when the transferor is the partner SIP set, describe here the procedure to
initiate such transfer

Test
Case Test Case N/A OK NOK Comment
Id
A B C
Transferee Transferor Transfer Target
Type of MCDU Type MCDU Type of MCDU
set (2) Number of set Number set (2) Number
(2)
OXE/Ext. OXEset- OXE/Ext. OXEset-
1 SIP SIPset-1
Call 1 Call 3
OXEset- OXE/Ext. OXEset-
2 SIP SIPset-1 OXE
1 Call 3
OXE/Ext. OXEset- OXEset-
3 OXE SIP SIPset-1
Call 1 3
OXEset-
4 SIP SIPset-1 OXE SIP SIPset-2
1
OXE/Ext. OXEset-
5 SIP SIPset-1 SIP SIPset-2
Call 1
OXE/Ext. OXEset-
6 SIP SIPset-1 SIP SIPset-2
Call 1
7 SIP SIPset-1 SIP SIPset-2 SIP SIPset-3

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5.9 Attendant

An attendant console (attendant set type : 4059EE, 4068…) is defined on the system. Call going to and coming
from the attendant console are tested.

Test
Test Case N/A OK NOK Comment
Case Id
1 Call to/from an attendant station

SIP set call to attendant

With SIPset-1, call attendant with prefix 9 (attendant

A
call prefix), attendant answers.

Check ringing back tone, display and audio.

Attendant call to SIP set

With the attendant station, call SIPset-1, SIPset-1

B
answers.

Check ringing back tone, display and audio.

nd
2 incoming call to SIP set while in conversation
with attendant.

SIPset-1 being in conversation with the attendant.

Make a PSTN incoming call to SIPset-1 and try to

answer the call.
C
SIPset-1 should not be able to take the second
incoming call. Check that the PSTN incoming call is
properly managed (the PSTN user should receive a
busy tone). Check that SIPset-1 stays in
conversation with the attendant station (SIPset-1 can
display an error message when it tries to put the
attendant station on hold).

Outgoing call to SIP set while in conversation

with attendant.

SIPset-1 being in conversation with the attendant.

With SIPset-1 try to call SIPset-2, check that this call
is not allowed.
D
SIPset-1 should not be able to put the attendant on
hold. Check that the SIPset-2 does not ring. Check
that SIPset-1 stays in conversation with the
attendant station (SIPset-1 can display an error
message when it tries to put the attendant station on
hold).

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Test
Test Case N/A OK NOK Comment
Case Id
Outgoing call to SIP set in “Do not Disturb”
(DND) state, by local feature if applicable

Enable DND on SIPset-1 and call it with the

E
attendant station.
Check that the call is not allowed, the ring back tone
and display
Cancel the DND on SIPset-1.
Outgoing call to SIP set in “Do not Disturb”
(DND) state, by system feature

Enable DND on SIPset-1 using 42 prefix (Do not

disturb prefix). After the voice guide, enter 0000
(<sip set password>). Wait for acknowledgement
F
from OXE.
Call SIPset-1 with the attendant station.
Check that the call is not allowed, the ring back tone
and display
Cancel the DND on SIPset-1 using 42 prefix (Do not
disturb prefix).

2 Call transfer from an attendant station

SIP set call to attendant, attendant transfers to

OXE set, semi-attended

With SIPset-1, call attendant with prefix 9 (attendant

call prefix), attendant answers.
A
From the attendant, call OXEset-1 and transfer
semi-attended.
Answer the call and check audio and display.

Redo the test with NwkSIPset-1.

SIP set call to attendant, attendant transfers to
OXE set, attended

With SIPset-1, call attendant with prefix 9 (attendant

call prefix), attendant answers.
B
From the attendant, call OXEset-1 and transfer
attended.
Check audio and display.

Redo the test with NwkSIPset-1.

OXE set calls to attendant, attendant transfers to
SIP set, semi-attended

With OXEset-1, call attendant with prefix 9

C
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer semi-
attended.
Answer the call and check audio and display.

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 Chapter 5 1 TESTS RESULT

Test
Test Case N/A OK NOK Comment
Case Id
OXE set calls to attendant (using attendant call
prefix), attendant transfers to SIP set, attended

With OXEset-1, call attendant with prefix 9

D
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer
attended.
Check audio and display.
External PSTN Call to attendant, attendant
transfers to SIP set, attended

E PSTN call to the attendant.

From the attendant, call SIPset-1 and transfer
attended
Check audio and display
SIP set call to attendant, attendant transfers to
External, semi-attended

From SIPset-1, call attendant with prefix 9 (attendant

F
call prefix), attendant answers.
From the attendant, call an external PSTN
destination and transfer semi-attended
Once in conversation, check audio and display.
OXE set calls to attendant (using attendant call
prefix), attendant transfers (semi attended
transfer) to SIP set in “Do not Disturb” (DND)
state (DND activated by local feature)

Enable DND on SIPset-1.

G
With OXEset-1, call attendant with prefix 9
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer semi-
attended.
Check that the transfer is not allowed
Cancel the DND on SIPset-1.
OXE set calls to attendant (using attendant call
prefix), attendant transfers (semi attended
transfer) to SIP set in “Do not Disturb” (DND)
state (DND activated by system feature)

Enable DND on SIPset-1 using 42 prefix (Do not

disturb prefix). After the voice guide, enter 0000
(<sip set password>). Wait for acknowledgement
H
from OXE.
With OXEset-1, call attendant with prefix 9
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer semi-
attended.
Check that the transfer is not allowed
Cancel the DND on SIPset-1 using 42 prefix (Do not
disturb prefix).

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 Chapter 5 1 TESTS RESULT

Test
Test Case N/A OK NOK Comment
Case Id
OXE set calls to attendant (using attendant call
prefix), attendant transfers (attended transfer) to
SIP set in “Do not Disturb” (DND) state (DND
activated by local feature)

Enable DND on SIPset-1.

I
With OXEset-1, call attendant with prefix 9
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer
attended.
Check that the transfer is not allowed
Cancel the DND on SIPset-1.
OXE set calls to attendant (using attendant call
prefix), attendant transfers (attended transfer) to
SIP set in “Do not Disturb” (DND) state (DND
activated by system feature)

Enable DND on SIPset-1 using 42 prefix (Do not

disturb prefix). After the voice guide, enter 0000
(<sip set password>). Wait for acknowledgement
J
from OXE.
With OXEset-1, call attendant with prefix 9
(attendant call prefix), attendant answers.
From the attendant, call SIPset-1 and transfer
attended.
Check that the transfer is not allowed
Cancel the DND on SIPset-1 using 42 prefix (Do not
disturb prefix).

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Chapter 5 1 TESTS RESULT

5.10 Voice Mail

Voice Mail notification, consultation and password modification must be checked.

MWI (Message Waiting Indication) has to be checked.

A voice mailbox is available for users SIPset-1, SIPset-2 and OXEset-1. The passwords and the voice mail
directory number are given in section “3.1 Hardware Configuration”.

For these tests, DTMF sending (RFC 2833) has to be validated in order to use Voice Mail menu.

Note 1: explicit subscription is required for RFC3842 MWI (ref 2.8.24.7 SIP Endpoint developers guide)

Test
Case Test Case N/A OK NOK Comment
Id

1 Message waiting indicator

Message display activation, MWI (1)

With SIPset-2 call the voice mail directory number.

Tested in the version
Follow the instructions in order to send a voice message
A 2.11.01 and found as
in SIPset-1 box.
fixed
Check that the MWI on SIPset-1 is activated.

2 Message listening

Message consultation

With SIPset-1 call the voice mail directory number.

Follow the instructions in order to listen your voice
Tested in the version
message leaved during the previous test. Check that
A 2.11.01 and found as
your can listen it and delete.
fixed
Check that MWI display is disabled on SIPset-1 after
message cancellation.

3 Voice mail management

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Chapter 5 1 TESTS RESULT

Test
Case Test Case N/A OK NOK Comment
Id
Password modification

With SIPset-1 call the voice mail directory number and

follow the Voice guide in order to modify the default
password.

When modification is accepted hang-up.

A
Recall the voice mail and try to log with a wrong
password. Check the rejection.

Recall the voice mail and try to log with the right
password. Check the service access.

4 Forward to voice mail

SIP call to a OXE user forwarded to Voice Mail

Forward the OXEset-1 to Voice Mail by dialing 51 prefix
(Immediate forward prefix). After the voice guide, enter
31300 (<Voice Mail number>).

A With SIPset-1 call OXEset-1 and check that you are

immediately forwarded to Voice Mail.
Check that you can leave a message

On OXEset-1 disable Voice Mail forwarding with 41

(Forward cancellation prefix).
OXE call to a SIP user forwarded to Voice Mail
Forward the SIPset-1 to Voice Mail by dialing 51 prefix
(Immediate forward prefix). After the voice guide, enter
31300 (<Voice Mail number>).

B With OXEset-1 call SIPset-1 and check that you are

immediately forwarded to Voice Mail.
Check that you can leave a message

On SIPset-1 disable Voice Mail forwarding with 41

(Forward cancellation prefix).

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 Chapter

6 6 Appendix A: SOLUTION DESCRIPTION

ALE’s latest Myriad Series includes 3 enterprise-class SIP phone models, the M3, M5, and M7,
suitable for executives, managers, and professionals with a high demand for quality telephony
services.
The Myriad Series delivers outstanding performance with enhanced HD audio quality, a modern
intuitive user interface, and enhanced telephony features. It enables users to easily manage their
calls, enjoy a smooth user experience and ultimately improve productivity.
Chapter 6 1 Appendix A: SOLUTION DESCRIPTION

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Chapter 6 1 Appendix A: SOLUTION DESCRIPTION

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 Chapter

Appendix B: PARTNER side

7
7

CONFIGURATION
To login into user profile.

Configuring IP Address on the Phone.

 Change Language：The default language is English. Menu Basic Setting Language，

Choose your language.
 The phone gets Dynamic IP address by default. If you need to set static IP address, you can choose
static IP mode for configuration.
 Static IP Settings as following：MenuAdvanced Setting（Default password: 123456)
Network IP ConfigIPV4 Setting
 Ipv4 Mode（select Static），then fill in the corresponding network configuration. Once IP Address is
configured, Phone gets reboot.

Using the Configured url http://10.200.2.4/ and enter username as “admin” and default
password as “Admin@123”.

Configure Network details information in the Network > IP Parameters Option.

Configure the SIP Account details in the Account > Basic settings.
Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

After Configuring the Network and SIP account settings. Phones starts to get reboots comes
inservice with “Registered” status.

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

Codec Configuration > Click on the Account > Codec.

Configure the Ringing tones in Settings option.

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

Configure DTMF Mode in Account > Advanced tab

Voicemail Configuration page.

Go to Account > Advanced tab > Configure Voicemail number and Message Indication number

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

Configure Forward options in Settings >Forward Options.

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

Configure DND option in Settings > DND Option.

Configure NTP server in Settings > Time

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

For Spatial Configuration: Click on the Account tab > Basic > Enter the “Main call server” IP
address in SIP server and “Backup IP address” in Secondary SIP Server option.

For PCS Configuration: Click on the Account tab > Basic > Enter the “Hostname of the Spatial
configuration” in SIP server and “PCS IP address” in Secondary SIP Server option.

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

To collect the logs, click on to the Maintenance > Logs Collection. Enter the server IP and port
details.

To capture the traces > click on the start option.

To configure the log levels, click on the Maintenance > Logs collection tab.

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

For Firmware Update: Click on the Maintenance > Binary Update

Firmware version details on the Webpage

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Chapter 7 1 Appendix B: PARTNER side CONFIGURATION

Firmware version details on the Phone.

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 Chapter

8 8 Appendix C: ALE side CONFIGURATION

Configurations done in OXE:

Launch OXE configuration application.

8.1 SIP gateway

Chapter 8 1 Appendix C: ALE side CONFIGURATION

8.2 SIP Proxy

8.3 Codec configuration

We can set the codec to G729 in the below path .

Select: System > Other System Param. > Compression Parameters

Compression Type Select: G729

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Chapter 8 1 Appendix C: ALE side CONFIGURATION

8.4 OXE Domain

8.5 SIP User Configuration

8.6 Licensing

Please check for the SIP licenses before the SIP phones are installed.

Please check the software lock 177 in SPADMIN

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 Chapter

9 9 Appendix D: SUPPORT PROCESS

The InterWorking Report describes the test cases which have been performed, the conditions of the testing and
the observed limitations.

This defines the scope of what has been certified and consequently what is supported by ALE.

If, even so, an issue is observed and is in the scope of the IWR, please open a Service Request using the
standard ALE process: https://businessportal.al-enterprise.com/technical-support-1

END OF DOCUMENT