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
    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 Quintum Tenor ASM200

Tenor AS VoIP MultiPath Switch
The Tenor ASM200, with two FXO/FXS and two VoIP ports, can be easily configured with Brekeke PBX to satisfy various dialing by automatically routing local calls from cellular phone and land lines to a VoIP network, and vice versa. This document contains basic information on how to setup Automatic Route Selection (ARS) to work with Tenor ASM200. For more information about this product, please visit


Quintum AS Series Installation

Please refer to the configuration documentation for instructions on installing Tenor ASM200 by visiting Using the link, below are some useful SIP configuration documents for the Tenor ASM200:

  • SIP Quick Start Guide
  • Tenor AS Quick Start Guide
Quintum AS Series Configuration

Please refer to configuration documentation for instructions on configuring your Tenor ASM200’s IP address/ subnet mask so that it is accessible from your network.

Using Quintum Tenor Configuration Manager

The Tenor ASM200 is managed by the Tenor Configuration Manager, a GUI Configurator, and Network Manager. You can obtain this software either by visiting the website or from the CD-ROM that comes with the unit. Please follow instructions carefully.

The following guide is for configuring the Tenor ASM200 to work with Brekeke PBX which allows Brekeke PBX users to make and receive PSTN calls. Please keep in mind that the settings listed below are only a guide to provide a basic understanding for users to setup Tenor ASM200 with Brekeke PBX. Should you need other parameters or settings, please consult the User Manual for this unit.

Open the Tenor Configuration Manager and click “Connect”.

Configuring Quintum AS Series for Brekeke SIP Server
1. SIP Signaling Group -1
[Primary SIP Server]: Brekeke SIP Server IP address
[Primary SIP Server Port]: 5060
[Primary Outbound Server]: Brekeke SIP Server IP address
[Primary Outbound Server Port]: 5060

[User Agent tab]: please note that user 3333 is created by clicking the “Add” button. You can enter any user ID that you want. Depending on which gateway product you are using, the Tenor ASM200 has two FXS port with which two Primary User account can be created.

2. CAS Signaling Group – line

Use the [Signaling Type] field to detect the end of a call. Depending on where you live, please use the drop-down menu and select the appropriate type that best fits your network environment.

Dialing to PSTN using “ One Stage Dialing ”
1. Hopoff Number Directory -1

This window contains telephone numbers for Brekeke PBX calls traveling over IP, and then into the PSTN. An ARS rule in Brekeke PBX will show where ARS rules are created to dial 9 + PSTN. You will also need to create a 9 in the Hopoff window as indicated below:

2. Trunk Circuit Routing Group – Line

[Forced Routing Number]: This field is where a SIP UA receives a call from the caller (in this scenario, calls will be routing to user 166).

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: 

Deploy Patterns: 
Yes No
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