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 Mediatrix 1204(FXO Gateway)

This document explains how to use Brekeke SIP Server with Mediatrix 1204(FXO Gateway). The Mediatrix 1204 is a high-quality and cost efficient VoIP gateway connecting IP networks to the PSTN. The Mediatrix 1204 connects up to 4 FXO trunks to an IP Ethernet access. The Mediatrix 1204 provides PSTN access for various VoIP endpoints such as IP phones, FXS devices, softphones and IP-based PBX and Key Systems. It is an efficient solution to maintain local PSTN breakout in remote locations that are converted to IP.http://www.mediatrix.com/

Configure Mediatrix 1204

Once the IP address is configured you can access Mediatrix 1204 FXO Gateway webpage by http://192.168.0.95/ (example of selected IP address).Please follow the following steps once the webpage opened.

User name: admin (default)
Password: 1234 (default)
Click on [OK] button
SIP Setting

The FXO Gateway will need to register to the SIP Server, so it needs to know the SIP Server’s IP address. Please follow the following steps to complete the configuration:

1. SIP Configuration Setting: SIP > Configuration

SIP Port: 5060 (Ex)
SIP Host:192.168.0.167 (Ex)
Proxy Port: 5060 (Ex)
Outbound Proxy Host: 192.168.0.167 (Ex)
Outbound Proxy Port: 5060 (Ex)
Click on [Submit] button
2. SIP Authentication Setting: SIP > Authentication

Index Realm: 192.168.0.167(Ex)
Username: 1204 (Ex)
Password:1234 (Ex)
Click on [Submit] button
Receiving Call setting

Set the user/phone number (1204 Ex) from which phone you want to receive the call in the “Automatic call address”. Please complete the following steps.

  1. Open the Unit Manager Network software which you can obtain from Mediatrix web site and discover the device
  2. Double click on the Telephony Attributes tab
  3. Set the “Automatic call” to Enable
  4. Automatic Call address: 1204(Ex)

With this setting, the user 1204 will receive the calls from PSTN through this gateway and Brekeke PBX.

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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