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

Setting Up a Skype Manager account

  1. Use your Skype Account to create a Skype Manager account at
  2. Create a SIP Profile from [Features] > [Skype Connect].
    When you have created a SIP profile, it will show your authentication information and proxy address with which you can use to set up ARS rules for Skype in Brekeke PBX
    such as:

    SIP User Password Skype Connect address Port
    99051000123456 xxxxxxxx 5060
  3. Purchase Calling channels at your SIP profile setting (There will be a monthly charge to your Skype Manager account.)
  4. To receive calls, you need to do one of the following, or both, depending on your planned usage:
    • From Caller ID, purchase an Online Number at your SIP profile, which you can use to receive calls from PSTN lines.
    • Assign Business Account as the destination for incoming calls from Skype.
  5. Allocate credit to your SIP profile to make/receive calls. (You must assign credit to your SIP profile both to make and receive calls with Skype)
Brekeke PBX ARS Configuration for Skype
  1. Create an ARS for Skype from Brekeke PBX admintool>[ARS]
  2. Set [General] fields as below:
    Register URI:
    Proxy Address:
    User: 99051000123456
    Password: xxxxxxxx
  3. Create a Pattern – IN as below:
    [Matching Patterns]
    To: sip:99051000123456@
    [Deploy Patterns]
    To: pbx_ext

    Select [Apply to Request URI instead of To] checkbox and save the setting.

  4. Create a Pattern – OUT as below:
    [Matching Patterns]
    To: sip:([0-9]{7,})@
    [Deploy Patterns]
    To: sip:$
  5. Click the [Save and update] button to register your Skype SIP account.
How to Make Calls
  • Incoming call (Skype -> Brekeke PBX, or to DID number assigned by Skype):
    • Calling from other Skype users to Brekeke PBX extension set in above ARS rule:
      Look for your Skype contact and click the [Call] button from Skype.
    • Calling from PSTN phone to Brekeke PBX extension set in above ARS rule:
      Dial the [online number] you purchased from Skype.
  • Outgoing Call (Brekeke PBX -> PSTN via Skype Out to PSTN line or Skype ID with online number):
    • Calling from Brekeke PBX extension to a PSTN number in the US through Skype:
      Dial 001 + area code + landline number
    • Calling from Brekeke PBX extension to other Skype users associated with online number:
      Dial the online number associated with the Skype ID as you would dial a PSTN line.
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