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Configure a Panasonic KX-NS SIP trunk

Trunking between a Panasonic PBX and GoTo Connect allows your current hardware to use their existing functionality, while adding some additional cross-platform functionalities with GoTo Connect phones.

Before you begin: You must add your device(s).
Note: It's recommended to start with a fresh base configuration—rather than using the initial NS700 wizard.

Prerequisites:

  • KX-NS700 version: 4.42025 or above.
  • One of the following supported DSP are installed inside system:
    • KX-NS5110 (1-10 trunks)
    • KX-NS5111 (11-30 trunks)
    • KX-NS5112 (31-100 trunks)
  • One of the following supported SIP Trunk Licenses are installed:
    • KX-NSM102
    • KX-NSM104
    • KX-NSM108
    • KX-NSM116
  • System is fully registered on Panasonic Partner Portal.

First, prepare your Panasonic PBX.

  1. Start the Panasonic Unified Maintenance Console.
  2. Select Connect, verify that you will be doing so at the INSTALLER level, and then choose the method of connection:
    • LAN
    • RS-232
    • Modem
    • USB to Connect to your PBX for interactive configuration
  3. Provide the IP address of the MPR card and the password of the INSTALLER Level.
  4. Select Connect.
  5. Login with the following information:
    Option Description
    Default IP 192.168.0.101
    Username INSTALLER
    Password 1234
  6. Verify that there is no pop-up for registration and verify the system version at top.
  7. From Network Service, select IP Address/Ports and complete the following settings:
    • IP address and Subnet Mask, default is 192.168.0.202/24.
    • Default Gateway IP address
    • DNS server address, 8.8.8.8 is preferred 8.8.4.4 is an alternative option
  8. From the left sidebar, select PBX Configuration > Configuration > Slot > Activation Key Status.
  9. Look at the Features in total column to verify the row IP Trunk (ch) shows at least 2 or higher and then select Apply > OK.
  10. From PBX Configuration, select Configuration > Slot and mouse over the Virtual Slot at the bottom of your screen.
  11. Select V-SIPGW16 at the top of your screen and drag it into Trunk slot 1.
  12. Hover over the newly added card and drag in into OUS so that configuration changes can be made.
    • Red dot: OUS
    • Green dot: INS
  13. Hover over the V-SIPGW16 card to select Shelf Property and then complete the following settings:
    Option Description
    NAT Traversal Off (Default)
    NAT Keep Alive Packet Sending Ability Enabled (Default is disabled)
    SIP Called Party Number Check Ability Disable Low to High (Default is High to Low)
  14. Select Apply > OK.

Second, configure the SIP Trunk.

  1. Hover over the V-SIPGW16 card to select Port Property and then complete the following settings:
    Option Description
    Channel Attribute Basic Channel
    Note: If using more channels, set the additional channel attributes to Additional Channel for...
    SIP Server Name <domain>.jive.rtcfront.net
    Note: Replace [domain] with your unique domain. PBX Administration (classic) users will find their domain in the parentheses at the top of their portal dashboard. GoTo Admin portal users will find their domain in Account Settings.
  2. Select Apply > OK.
  3. From Port Property > Account, complete the following settings:
    Option Description
    User Name Provided by GoTo Connect
    Authentication ID Provided by GoTo Connect
    Authentication Password Provided by GoTo Connect
  4. From Port Property > Register, complete the following settings:
    Option Description
    Register Sending Interval 300 seconds (Default is 3600)
  5. Select Apply > OK.
  6. From Port Property > Calling Party, complete the following settings:
    Option Description
    Header Type P-Preferred-Identity Header (Default - From Header)
    From Header - User Part PBX-Clip
  7. Select Apply > OK.
  8. FromPort Property > Calling Party, complete the following settings:
    Option Description
    Type Request-URI (Default is To Header)
  9. Select Apply > OK.
  10. From the left sidebar, select PBX Configuration > CO & Incoming Call > DID Table to complete the following settings:
    • Enter the DID numbers or phone numbers of the account.
    • Enter a valid destination for day, lunch, break, and night.
    • Enter extension numbers or group numbers.
  11. Select Apply > OK.

Add per extension outgoing caller ID (optional)

  1. From the left sidebar, select PBX Configuration > Extension > Wired Extension > Extension Settings > Clip.
    Tip: You can also do this for Portable Stations.
  2. Select Add CLIP. This is the caller ID that you wish to show for each extension (default blank).
    You can also add a CLIP ID (blank by default) for the Incoming Call Distribution Group ( Group Settings > ICD Group Button).

Lastly, activate and test the SIP Trunk.

  1. From PBX Configuration > Configuration > Slot, hover over Virtual Slot and the V-SIPGW16 card to select INS.
    You should see a green indication on the card.
  2. Verify Port Property Connection column reads INS, to ensure it is connected properly.
    You may need to OUS and INS a couple of times if it’s the first time connecting to GoTo Connect Network.
  3. Select Apply > OK.
Related Articles:
  • Configure Cisco Call Manager SIP trunk with Adtran SIP header manipulation
  • Configure a Panasonic KX-KDE/NCP SIP trunk

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