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 DX Gateway

Gateway Tenor DX VoIP MultiPath/Gateway Switch
The Tenor DX gateway is T1/E1 Gateway, it 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 DX Gateway. For more information about this product, please visit


1. Quintum DX Series Installation.

Please refer to the configuration documentation for instructions on installing Tenor DX gateway by visiting


2. Quintum DX Series Configuration

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


3. Using Quintum Tenor Configuration Manager

The Tenor DX Gateway 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 DX Gateway 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 DX Gateway 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”.

4. Configuring Quintum DX Series Gateway for Brekeke SIP Server.

Digital Line Configuration-General:

Choose one of the ports connects to Brekeke PBX and another port connects to PSTN.
Please go to Basic Config > Digital Line Configuration > General

Digital Line Configuration-Protocol:

Use the [Signaling Protocol] 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.
Please go to Basic Config > Digital Line Configuration >Protocol

CAS Signaling Group-1 General:

Configure the CAS signaling, which is depended on Digital Line Configuration-Protocol, please see above.
Please go to Advanced Explore > Circuit Configuration > CAS Signaling Groups > CAS Signaling Group-1 > General

CAS Signaling Group-1 Signaling:

Configure the Incoming/Outgoing Protocol, which is depended on what kind of E&M signaling the PSTN has
Please go to Advanced Explore > Circuit Configuration > CAS Signaling Groups > CAS Signaling Group-1 > Signaling

SIP Configuration:

[Primary SIP Server]: Brekeke SIP Server IP address
[Primary SIP Server Port]: 5060
[Register Expiry Time]: 3600Brekeke SIP Server IP address
[UserID]: 2001(e.g.)

Please go to basic config > SIP Configuration

Once you save the setting, and reboot the gateway.

Go to the Registered tab of Brekeke SIP Server admintool.

5. Call Setting

Voice Codec-1 or 2

Modify the the voice codec that Brekeke PBX is acceptable.
Please go to [Advanced Explore] > [VoIP Configuration] > [Voice Codecs]

Making Call to PSTN using “One Stage Dialing”:

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:
Please go to Advanced Explore > Circuit Configuration > Trunk Routing Configuration > Hopoff Number Directory-1

Receiving the call from PBX

[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 119).
Go to [Advanced Explore] > [Trunk Routing Configuration] > [Trunk Circuit Routing Groups] > [Trunk Circuit Routing Group-1] > [Advance]

Go to [Advanced Explore] > [Circuit Configuration] > [Line Circuit Routing Groups] > [Line Circuit Routing Group-1] > [Advance]


6. 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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