Interoperability

  1. SIP Phones
    1. How to set paging function on the phone side
    2. SIP softphone we recommend
    3. Hard Phones
      1. ClearOne MAX IP Conferencing Phone
      2. Grandstream Networks Budge Tone - 100 Phone
      3. inter-tel
      4. ipDialog SIP Tone III Phone
      5. ipDialog SipTone™ V Phone
      6. Linksys IP Phone
      7. Nokia Phone
      8. Panasonic KX-TGP550 T04 DECT
      9. Panasonic KX-UT1xx
        1. Configure shared line in Panasonic KX-UT1xx
        2. Set up Flexible Button on Panasonic KX-UT1xx
      10. Polycom Kirk DECT
      11. Polycom SoundPoint IP 650/330 Phone
      12. Polycom SpectraLink 8020 Wireless Telephone
      13. Polycom VVX600 Business Media Phone
      14. Polycom VVX Intercom feature
        1. Intercom with provisioning
      15. SpectraLink 8400 Wireless Telephone
      16. SAXA IP Netphone SX Phone
      17. Snom m3 VoIP wireless DECT phone
      18. Snom Phones
      19. Swissvoice IP10S Phone
      20. Ubiquiti UniFi VoIP Phones
      21. Vtech Business Phones
      22. Yealink SIP Phones
      23. ZyXEL Prestige 200w
    4. Soft Phones
      1. Age Phone
      2. Bria
      3. Kapanga
      4. X-Lite
  2. VoIP SIP Gateways
    1. AddPac AP1200
    2. AudioCodes Mediant 1000(FXO/FXS)
    3. AudioCodes MP-104 (FXO)
    4. AudioCodes MP-114 (FXO/FXS)
    5. AudioCodecs TP-260
    6. Cisco Systems ATA186
    7. Grandstream Networks HondyTone 486
    8. Mediatrix 1104(FXS)
    9. Linksys SPA-3000
    10. Mediatrix 1204 (FXO)
    11. Medicatrix 2102 (FXS)
    12. Quintum Tenor ASM200
    13. Quintum Tenor DX Gateway
    14. Sangoma NetBorder Express Gateway
    15. TAINET - Venus 2908
    16. Yamaha RTV 700
    17. Yamaha RTX 1000
    18. ZyXEL Prestige 2602 HW
  3. ITSP
    1. BroadVoice
    2. DIDx
    3. iConnectHere
    4. InPhonex
    5. Megapath
    6. sipgate
    7. Skype
    8. Triad Telecom
    9. Virtual GlobalPhone
    10. VoIP INNOVATIONS
    11. Voxbone
    12. VoIPUSER
    13. Vonage
  4. WebRTC client
    1. JsSIP
  5. Misc
    1. Application-Level Gateway(ALG)
      1. InGate SIParator 4.6.1
        1. Standalone SIParator, with SIP server on the outside
        2. DMZ SIParator, with Brekeke SIP Server on the outside
    2. Ascom AA60
    3. Avaya PBX (CM 5.2)
    4. CyberData VoIP Ceiling Speaker
    5. CyberData VoIP Intercom
    6. DTH VoIP Billing
    7. Microsoft Lync
    8. Microsoft Lync or OCS
      1. Brekeke SIP Server and Lync or OCS
      2. Brekeke PBX and Microsoft Lync or OCS
        1. Configure Brekeke PBX for Lync or OCS
        2. Configure Lync or OCS for Brekeke PBX
    9. Microsoft Speech Server
    10. Mobotix T24M-Sec-D11 Hemispheric IP Video Door Station
    11. Nortel CS1000
    12. Radius Server
      1. Jerasoft Development - JeraSoft VCS
      2. Jerasoft Development's VCS Dynamic Routing
      3. Clearbox Radius
      4. FreeRADIUS
    13. Jeron's Nurse Call System - Provider 790
    14. Rauland's Nurse Call System - Responder 5
    15. Sangoma NetBorder Call Analyzer

Connecting with Sangoma NetBorder Express Gateway

This document contains basic information on how to setup Automatic Route Selection (ARS) to work with Sangoma NetBorder Express Gateway.
For more information about product, please visit
Sangoma.com
To install Sangoma NetBorder Express Gateway, please refer to the user guide for system requirements and installation instructions about board and software.

 

Configure Sangoma NetBorder Express Gateway

The default port where the Netborder Express web configurator listens for connections is 7783. Connect to the Web interface with http://localhost:7783/

 

Routing rules

Configure routing rules to send inbound calls to a Brekeke PBX user Such as: PBX IP is 192.168.0.89 and calls will be routed to pbx user 1000, then set as below

<rule name="default_sip_out" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="false">
<condition param="pstn.in.channelName" expr=".*">
<condition param="pstn.in.ani" expr="(.*)">
<out_leg name="" media_type="sendrecv">
<param name="sip.out.requestUri" expr="sip:1000@192.168.0.89:5060" />
<param name="sip.out.from.uri" expr="sip:%0@GW_HOST_IP:GW_SIP_PORT" />
<param name="sip.out.from.displayName" expr="Netborder Express Gateway" />
<param name="sip.out.transport" expr="udp" />

 

Dial Plan or ARS Setting

For Brekeke SIP Server Users, please check wiki post Connecting Gateways with BSS

For Brekeke PBX users, please check wiki post Connecting Gateways with BPBX or Update Issues from v2.2.7.7 and before
Sample ARS Pattern-OUT for outbound calls at [Brekeke PBX Admintool]>[ARS]>[New Route]

Pattern – OUT

----------------------- 
Matching Patterns: 
To=sip:([0-9]{7,25})@

Deploy Patterns: 
To=sip:$1@gateway_IP
------------------------
Notes
  1. If host system has more than one IP addresses, you need to either assign only one IP address to the pc
    or set ” netborder.net.primaryIPAddress=192.168.0.89 ” in Netborder_installation_directory/config/gw.properties
  2. To pass the DID to Brekeke PBX
    please refer to http://wiki.sangoma.com/wanpipe-windows-nbe-appendix#did for settings
  3. The DTMF has to be sent in the rtp stream as described by RFC 2833
    To enable RFC 2833 DTMF support, go to netborder_installation_directory/config/gw.properties
    uncomment “netborder.media.rtp.rfc2833Supported=true”
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