NSX Series V5.

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- Summary -
Rev 1.0 NOV. 2020
 Table of Contents for V5.3

No. Contents NSX

Version 5.3
1 Meet Me Conference Yes
2 UCMA Mobile Softphone Yes
Easy Installation
3 SIP Trunk Easy Set Up Yes
4 Static Routing Yes
5 CA V6 / IP Softphone V5 – Remote Yes
Working improvements
6 CLIP for emergency dial Yes

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 NSX ExpansionGW: Simplified Isolated mode

For NS Series systems operating as an Expansion Gateway of an NSX solution, if the NS

system goes to Simplified Isolated mode then only basic features are supported.

Unsupported features
NS version 7.0 features
Unified Messaging
are supported except the following:
Simple Voicemail (SVM)
CTI / CSTA Interface
Built in ACD Reporting
UCMA Mobile Softphone

(NOTE: UCMA Mobile Softphone users should be registered to NSX main unit only).

Following new features are supported Supported features

in Simplified Isolated Mode: KX-DT635 / KX-DT680 Digital PT
CLIP for emergency dial
Emergency dial alert to PT
SIP trunk static routing

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Overview
 Meet Me Conference

Start conference by calling a dedicated Virtual-PS extension configured as a conference

room number and including a PIN code.
• Up to 48 Conference Rooms available.
• Not necessary for Host to start conference before participants join.
• Virtual-PS extension number is the Conference Room Number.
• Internal parties dial Conference Room number directly.
• External parties dial in through DISA attendant.

PSTN

Max. 32 parties per

Conference room

Virtual-PS: 300

Virtual-PS 51-98 can be

used as Conference room

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 Meet Me Conference

Operation

Example Virtual-PS extension number (Conference Room number): 300 PIN: 12345678

1) Internal participants: Dial “300” then “Enter PIN” is shown at PT display + Special tone is heard: dial
Conference PIN: “12345678” (SIP devices should enter all digits before calling - 30012345678)

2) External participants: Dial DDI number for DISA Attendant, Dial the Conference Room number
followed by PIN (cannot use DISA key options to transfer to Conference Room)

Virtual-PS: 300
PSTN PIN: 12345678

DISA

During Conference, PT display shows:

“300 : ConfRoom1 01”
Conf room number > name > number of participants

V-PS Extension name presented as Conf room name.

If no name configured then only number is displayed.

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 Mobile Softphone Easy Installation

PBX can send KX-UCMA configuration to each user by e-mail.

Configuration can also be adjusted manually after initial set up.

Multiple SIP extensions can be selected.

Basic SIP extension settings are sent by email.

Configuration is attached to e-mail by CSV and QR.

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 SIP Trunk Easy Set Up

Recommended settings for SIP provider are configured automatically when V-

SIPGW card is installed.
• 1 Basic channel and 3 additional channels are also installed automatically.
• ID and password should be assigned manually.
• Other settings such as static routing may be assigned manually depending on carrier

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 Static Routing

IP data for different destinations can be directed through different router gateways.
This also applies to VoIP data by DSP.
IP Address > Routing page

SIP Trunk traffic through different router

to all other remote IP traffic.

SIP Carrier Network

internet
172.16.1.1
VPN or MRG
172.16.2.3

192.168.0.99 192.168.0.1
Default Gateway
Web-MC Programming:
192.168.0.151 192.168.0.152 192.168.0.101 IP Address = 192.168.0.101
Subnet mask = 255.255.255.0
Default GW = 192.168.0.1
PBX
Static Routing for specific network
172.16.1.1 via 192.168.0.99

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 CA V6 / IP Softphone V5 – Remote Working

Following improvements made to CA and IP softphone for PC:

- TLS encrypted communication with remote CA clients and IP Softphone (VPN-less)
- SRTP Encrypted VoIP with remote IP Softphone.
- CA Chat over remote connection without VPN (via PBX)

Head office
www
ROUTER

home / remote office

ROUTER

Home

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 CLIP for Emergency dial

When an emergency number is dialed, “CLIP for Emergency Dial” configured for the device
location is sent to the CO trunk as the Calling Party ID (SIP or ISDN only).

192.168.0.192 to 192.168.0.254 / CLIP: 03330004444

4th floor Emergency Call Centre
CS
192.168.0.128 to 192.168.0.191 / CLIP: 03330003333
3rd floor PBX
CS
192.168.0.64 to 192.168.0.127 / CLIP: 03330002222 Location base CLIP
03330004444
e.g. Database Address:
2nd floor 03330004444 XYZ street 12345
4F East Location
192.168.0.1 to 192.168.0.63 / CLIP:03330001111
1st floor

*When “CLIP for Emergency Dial” is not programmed, CLIP of extension is sent.

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 Call Back from the Emergency Call Centre

When the emergency call Centre calls back, the call is automatically directed by the PBX to the
terminal that made the outgoing call to the emergency call Centre.
(The call is not routed by standard DDI programming in this case).

Emergency Call Centre

Make Emergency Calls back to

call with CLIP 0202-123-4504
0202-123-4504

“Emergency Dial Alert log” PBX Routes the call based on the
Caller / To / CLIP
PBX Emergency Dial Alert log, not DDI table
102 / 112 / 02021234504

102

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Programming
 Meet Me Conference

1) Enable System Option bit 23_5 (displayed under Option 99 tab)

Virtual-PS locations 51-98 can then be used for Conference room facility.

2) Portable Station extension number as 3) Edit User – User PIN is conference room PIN code.
conference room number. (Name is displayed on PT display).

ConfRoom300

ConfRoom300
300 ConfRoom300

• Extension (internal) or DISA (external) user dials the room number and PIN code to join the meeting.
• Meeting room numbers 1 to 48 use the extension number of Portable Station ports 51 to 98.

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 Meet Me Conference

4) Using Sales Company login, Enable System Options > Reserved bit 24_3 for PT display
to also show Conference room quantity of participants.

Quantity of participants in conference

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 Meet Me Conference

Feature Limitations / Conditions

• If DISA security mode is set to ALL, Walking COS must be used to allow access to Conference Rooms.
• TCA DECT devices cannot be registered to ports 51-98.
• PS extension number set for Conference Room cannot be used for any other purposes.
• PS extension Call Forwarding / Intercept do not work for these extensions.
• Analogue CO trunks are not supported.
• Extension with held call or mid-transfer cannot enter conference room.
• If only 1 participant is connected, they are disconnected automatically after 10 minutes.
• All participants are automatically disconnected after 3 hours.
• If PIN code is modified during active Conference, new participants must use the new PIN code to join.

DSP Resources
• DSP resources are required for all connected IP extensions, IP Trunks and Conference Room participants.
• DISA attendant will only use DSP resource during transfer to conference room.
• Max. 32 participants per Conference Room
• Max. 256 participants total in conference rooms with DSP installed.
• Max. simultaneous conference participants is limited by the DSP resource available.

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 Mobile Softphone Easy Installation
1) PBX needs working SMTP email configuration

4) Select the target extensions for setup data:

2) Mobile Softphone user needs valid e-mail address

3) Mobile Softphone SIP extension port needs valid

Extension number and Password

SIP extension port must

be OUS or Fault status

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 Mobile Softphone Easy Installation
Email received at Mobile Softphone user side:

Up to 32 characters

Up to 256 characters

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 Mobile Softphone Easy Installation

Parameters configured by e-mail CSV or QR Code configuration:

Same
value
If “NAT – SIP-TLS Server Port No. =0” then
“NAT – SIP Proxy Server Port No.” is sent

TLS is recommended for increased security

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 Mobile Softphone Easy Installation
Conditions

When Mobile Softphone is Sub-Device

then e-mail will be sent to address
assigned to main extension user.

E-mail for Mobile Softphone set up is not displayed in “E-mail Report” log

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 SIP Trunk Easy Set Up

Initial settings for each SIP provider can be set automatically for V-SIPGW card installation.

Common settings are deployed as per setup guidance

documents at Partner Portal or global support web site:

https://panasonic.net/cns/pcc/support/pbx/manual/kx-ns/sip-trunk/index.html

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 SIP Trunk Easy Set Up

“NAT Traversal” default values

Shelf Property > Virtual SIP Gateway

NAT Traversal Table 1 2-7 8

Default Value of
Off
Fixed IP Addr. UPnP@ Default Values of tables 2-8
“NAT Traversal” (Off : before v5.3) (Off : before v5.3) are changed from V5.3

Port Property - Virtual SIP Gateway

Select NAT Traversal Table from 1 to 8 to use as required

by carrier setup guidance.
1: Off
2-7: Fixed IP Addr.
8: UPnP@

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 Static Routing

Individual IP Gateways can be used for different IP subnet destinations. E.g. multiple SIP trunk
carrier services, internet access for remote maintenance or remote IP extensions.

1) Enable a routing table entry

2) Enter the Destination IP Subnet

address and Netmask for the
target IP network.

3) Enter router Gateway Address

(must be on same network as NSV
LAN IP address)

4) OK/Apply to complete the routing.

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 CA V6 – Remote Working Improvements

Bit Default Specification when the bit is enabled

23-6 Enabled * • TLS applied to Panasonic CA. Chat also works via PBX instead
of P2P for remote users without VPN.
• CA client must be version 6.0 or later
(NOTE: external CA Server software not supported on NSX)

* If existing PBX is upgraded, 23-6 will be “Disabled”. If initialized at v5.3 then default is “Enabled”

Built-in CA Server port number needs to be configured for port forwarding/open port at PBX side router/firewall.

There is no other special additional configuration required for CA over TLS connection.

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 CA V6 – Remote Working Improvements

CA V6 client must also be installed for these features to be supported and operate correctly.

PBX Firmware Version CA Client Version Specifications

V5.x (or earlier)
V5.2x (or earlier)
V6.x (or later) CA works but no encryption or
V5.x (or earlier) remote chat feature
Existing/configured system, updated
to V5.3x (or later) V6.x (or later)
V5.x (or earlier) CA does not work at all *
New or factory initialized system with
V5.3x (or later) CA with TLS encryption and remote
V6.x (or later)
CHAT feature supported

* CA Client must be upgraded to V6.x (or later)

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 IP Softphone V5 – Remote Working Improvement

IP Softphone V5 MGCP-TLS/SRTP

There is no additional set up required for IP Softphone MGCP-TLS/SRTP operation.

Configuration as per NT6 series MGCP-TLS/SRTP MRG connection.
Set the IP-PT Extension Ports > Location > Phone Location to Remote + Local for all remote extension
devices.

Set MGCP-TLS/SRTP to Enable for encrypted communication, if required .

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 CLIP for Emergency dial

“CLIP for Emergency Dial” can be programmed according to IP address range for IP terminal and
according to device port for legacy device.
(IP terminal means KX-NT Series phones, IP Softphone, any SIP phone and DECT on IP-CS).

- Up to 64 Network Addresses can be configured with dedicated CLIP as location identity for Emergency Dial

- Each Legacy device can have CLIP as location identity for Emergency Dial

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 Call Back from the Emergency Call Centre

- This feature applies to any Numbers configured in the Emergency Number table:

If selected, then when an emergency number

is dialed, FWD All/DND is disabled
automatically and call waiting is enabled
automatically.

If FWD Busy/NoAns/Busy&NoAns is already

configured, then settings are not changed or
disabled.
112

- If <DDI number received from ISDN/SIP network> and <CLIP sent to ISDN/SIP network> are not identical
then call back does not work. DDI number should be modified to match CLIP by programming.

DDI 4567 can be modified to

2021234567 for example.

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 Call Back from the Emergency Call Centre

• Maximum 20 emergency calls are logged.

• Caller information and CLIP are recorded in the log.
(EXT recognized as physical port number. CLIP logged in case it is CLIP for Emergency Dial)
• Older log is overwritten automatically when 21st emergency call is made.
• “Alert & Log” must be enabled.

Condition
- Call back can reach caller directly by DDI when <DDI number received from ISDN/SIP network for call back> and
<CLIP sent to ISDN/SIP network when making an emergency call> are identical.

- When the calling extension/user has two or more devices (e.g. using One Number, Wireless XDP etc.) all devices
ring with the call back.

- With the “Cancel FWD/DND” option enabled, when an emergency number is dialed (regardless of “CLIP for
Emergency Dial” setting), FWD/DND is disabled automatically and call waiting is enabled automatically.
If they need to be enabled/disabled for normal operation then settings must be changed manually.

- If <DDI number received from ISDN/SIP network> and <CLIP sent to ISDN/SIP network> are not identical then call
back does not work. DDI number should be modified to match CLIP by programming.

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END