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 Linksys SPA-3000

The SPA-3000 is a 1 port FXO + 1 port FXS analog telephone adapter. You can configure this device to work with Brekeke PBX to leverage your broadband phone service connections by automatically routing local calls from cell phone and land line to VoIP service provider and vice versa. This document will explain how to set up the Automatic Route Selection (ARS) for the SPA-3000. For more information on SPA-3000, please go to Linksys company website.

SPA-3000 installation

Please refer to section “How to initially set-up an SPA-3000?”

Initial SPA Configuration

Please refer to section “How can I configure the SPA-3000?”

Using a Web Browser to Enable Service

The SPA-3000 has the Web Configuration that allows you to easily view and manage your gateway’s settings through an internet browser. Please follow the instructions below carefully.

  • Run a web browser application on the same network that the SPA-3000 is connected.
  • Open a session to the SPA-3000 by typing: http://[spa ip address]/admin/advanced where the is the ip address (without the [ ]) from the DHCP or manually entered from Initial SPA Configuration. For example (http://192.168.0.106/admin/advanced)
  • The following is a guide for configuring the SPA-3000 to work with Brekeke PBX to allow you to make and receive VoIP calls. Please note that only required parameters, within the Brekeke PBX and SIP server domain, are discussed here. Should you need to understand other parameters or settings, please consult the Userfs Manual of this product.
1. PSTN Line Tab

The PSTN LINE tab is where you configure NAT Settings, Network Settings, SIP settings, Proxy and Registration, and Subscriber Information, Audio Configuration, Dial Plans, VoIP-To-PSTN Gateway Setup, VoIP Users and Passwords, PSTN-To-VoIP Gateway Setup, FXO Timer values, PSTN Disconnect Detection, International Control.

The sections where you need to check or change are the SIP Settings, Proxy and Registration and Subscriber Information, Audio Configuration, Dial Plans, VoIP-To-PSTN Gateway Setup, and PSTN-To-VoIP Gateway Setup.

  • [SIP Settings]->[SIP Port] : 5061 or whatever is appropriate for you.
  • [Proxy and Registration]->[Proxy]: The IP address of the machine where Brekeke SIP Server is running.
  • [Subscriber Information]->[Display Name]: The name you want to be identified for caller id.
  • [Subscriber Information]->[Username]: Username for the account to be registered for Brekeke SIP Server.
  • [Subscriber Information]->[Password]: The password for the user for authentication purposes.
  • [Audio Configuration]->[ Preferred Codec]: G711u (or what is supported by the device)
  • [Dial Plans]->[Dial Plan1]: PBX extension that receives incoming calls
  • [VoIP-To-PSTN Gateway Setup]->[VoIP-To-PSTN Gateway Enable]: yes
  • [PSTN-To-VoIP Gateway Setup]->[PSTN-To-VoIP Gateway Enable]: yes
  • [PSTN-To-VoIP Gateway Setup]->[PSTN Ring Thru Line 1]: no
2. Line 1

  • [SIP Settings]->[SIP Port]: 5060
  • [Proxy and Registration]->[Proxy]: The IP address of the machine where Brekeke SIP Server is running.
  • [Subscriber Information]->[Display Name]: The name or number that you want to be identified on a caller id feature.
  • [Subscriber Information]->[User ID]: The user id to be registered with Brekeke SIP Server.
  • [Subscriber Information]->[Password]: The password for the user for authentication purposes.
3. One-stage or two-stage dilaing

3.1. One-Stage Dialing

  • Settings at SPA 3000 for One-Stage Dialing:
    [VoIP-To-PSTN Gateway Setup]->[One Stage Dialing]: yes
  • To make a call:
    Dial PSTN phone number of the destination + “#”

3.2. Two-Stage Dialing

  • Settings at SPA 3000 for Two-Stage Dialing
    [VoIP-To-PSTN Gateway Setup]->[One Stage Dialing]: no
  • To make a call
    Dial 444# and wait for the dial tone, where: 444 is the value that was set in the “User ID” field)
    Dial phone number of the destination

4. 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

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Matching Patterns: 
To=sip:([0-9]{7,25})@

Deploy Patterns: 
To=sip:$1@gateway_IP
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