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. VoIP INNOVATIONS
    10. Voxbone
    11. VoIPUSER
    12. 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

VoIP INNOVATIONS

VoIP INNOVATIONS (https://voipinnovations.com/) offers telephone service using the VoIP standard SIP. The service allow users making and/or 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 VoIP INNOVATIONS.

 

Setting Up a VoIP INNOVATIONS Account

After log in your account, purchase DIDs.

Example: DID number “16501234567” is used as an example.

DID VoIP INNOVATIONS primary origination server Password
16501234567 64.136.173.31 N/A *1

 

ARS Configuration at Brekeke PBX for VoIP INNOVATIONS

Click [ARS] menu in Brekeke PBX Admintool, then click on [New Route] button and set:

[Route Name] "16501234567" or any other name 
[Tenant] Set the tenant name that uses this route. *2
[Register URI] sip:+16501234567@127.0.0.1 *3,*4

 

For Inbound call:

Create new Patterns – IN at this new route, and set:

[Apply to Request URI instead of To] Checked
[Apply only to calls related to registration] Checked
For Outbound call:

Create new Patterns – OUT at this new route, and set:

Matching patterns
[To] sip:([0-9]{10,20})@

Deploy patterns
[From] sip:16501234567@
[To]  sip:&t1@64.136.173.31

 

*1. Authentication is not required.

*2. Brekeke PBX Multi-tenant edition Only.

*3. Set “sip:<your DID number>@127.0.0.1″.

*4. VoIP INNOVATION send calls with the Plus”+ and Country Code(CC) as default. You can strip them at VoIP INNOVATIONS’ website.  If you strip them , you should change the value of [Register URI] like “sip:6501234567@127.0.0.1“.

 

 

Example of settings:

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