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 Triad Telecom

Triad Telecom(http://triadtelecom.com/ ) is the nation’s leader in business class voice over IP telecommunications service and VoIP trunking. The service allows users making/receiving VoIP calls to/from VoIP telephone numbers and making VoIP calls to PSTN telephone numbers. This document will explain how to utilize Brekeke PBX with Triad Telecom.

Setting Up a Triad Telecom Account

Triad Telecom authorizes client calls by client IP address. There is no registration required from clients to Triad Telecom.
Please obtain a DID number from Triad Telecom website for PSTN calls.

DID/PSTN Telephone Number Triad Telecom Server Address
650-227-0001( Ex) 208.70.21.21
Brekeke PBX ARS Configuration for Triad Telecom

Create ARS patterns IN and OUT for sending/receiving calls by Triad Telecom from Brekeke PBX Admintool > [ARS] menu > [New Route]. Set up ARS route as following:

[General] section:
This setting will register your DID number to Brekeke PBX bundled SIP Server.
And default dial plan “To PBX From ITSP” will be applied to route the calls from Triad Telecom to PBX side.

[Register URI] sip:&v1@127.0.0.1
[Proxy Address] 127.0.0.1

[Patterns – IN] section:
To route the calls from Triad Telecom to a Brekeke PBX user which is set in v3, such as an auto attendant number.

-------------------
[Matching Patterns]
To: sip:&v1@

[Deploy Patterns]
To: &v3
-------------------

[Patterns – OUT] section:
10-digit dialing number calls will be considered as outbound calls and sent to Triad Telecom server IP.
10-digit PSTN number is required for all US domestic calls with Triad Telecom.

-------------------------
[Matching Patterns]
To: sip:([0-9]{10})@

[Deploy Patterns]
From: "&v1"<sip:&v1@Brekeke_PBX_global_IP>
To: sip:$1@208.70.21.21
--------------------------

[Edit Variables]:
click on [Edit Variables] at upper-right of ARS template page and enter the information for each variable.

  1. Enter the user ID that your received from Triad Telecom in “v1”: 6502270001(ex)
  2. Enter the PBX Extension number for receiving incoming calls in “v3”:1001(ex)
  3. Click on [Save And Update] button.
  4. Go back to Route Template page. Uncheck [disabled] button in General setting.
  5. Click on [Save And Update] button.
How to Make Calls
  • Incoming call: Calling from PSTN phones in the US to Brekeke PBX extension 1001.
    Dial as: 6502270001 or 16502270001
  • Outgoing Call: Calling from Brekeke PBX extension to a PSTN number in the US.
    Dial as: Area code + Landline number (a 10-digit destination PSTN number)
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