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 AudioCodes Mediant1000

The Mediant1000 is 4-port FXS and 4-port FXO telephony Media Gateway. 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 use Brekeke PBX with AudioCodes Mediant1000(FXS/FXO Gateway). For more information on this product, please go to AudioCodes Company Website.

Configure AudioCodes Mediant1000

Once the IP address is configured you can access AudioCodes Mediant1000 FXS/FXO Gateway webpage by (example of selected IP address).Please follow the following steps once the webpage is open.

  1. Enter User name: Admin (default)
  2. Enter Password: Admin (default)
  3. Click on [OK] button

Note: the username and password are case-sensitive.

Brekeke SIP Setting

Since the Mediant1000 is 4-port FXS and 4-port FXO gateway, the FXS Gateway will need to register to the Brekeke SIP Server, so it needs to know the Brekeke SIP Server’s IP address, Please follow the following Steps to complete the Configuration.

1. Quick Setup

This is where you set up the IP addresses and SIP parameters.

  1. Enter the “IP Address”:
  2. Enter the “Subnet Mask”:
  3. Enter the “Default Gateway IP Address”:
  4. Choose the “Working with Proxy”: Yes
  5. Enter the “Proxy IP Address”:
  6. Enable Registration
  7. Click on [Reset] button
FXS Gateway Setting
1. End Points’ Phone Numbers

Select End Point Phones’ Number menu. Map each channel to the phone number which is PBX User Extension Number.

2. Brekeke SIP Server’s Registration Page

Click the [Registered Clients] tab of Brekeke SIP Server admintool.

All ports of the AudioCodes gateway are registered with Brekeke SIP Server.<

FXO Gateway Setting
1. Automatic Dialing

Select Endpoint Settings > Automatic Dialing menu. Map FXO port to the destination phone number from which phone you want to receive the call.

2. Channel Select Mode

Select Protocol Management > Protocol Definition > General Parameters menu. Change your channel settings parameters here.

  1. Select “By Dest Phone Number” when you use FXS gateway and two stage dialing
  2. Select “Descending” when you use one stage dialing.
3. 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 BPBXor Update Issues from v2.2.7.7 and before

Sample ARS Pattern-OUT for making outgoing calls through FXO

Pattern – OUT:

Matching Patterns 

Deploy Patterns

3.1 One Stage Dialing

The image below shows the settings for One-Stage Dialing.
To make a call:Dial PSTN line phone number of the destination

3.2 Two Stage Dialing

The image below shows the settings for Two-Stage Dialing.
To make a call:Dial 1201# and wait for the dial tone.
(Where 1201 is the value that was set in the User name field (see 3.1)).
Dial phone number of the destination.

Current Disconnect Duration Setting

Disconnect signal from PSTN company is different depending on the country. Some countries use Polarity Reversal and some use Current Disconnect. The following setting is an example of United States.

If the call is not immediately disconnected, please do the following steps:

  • Go to Advanced Configuration > Configuration File
  • Click on [Get ini File], you will receive “BOARD.ini” file
  • Once you open “BOARD.ini” file, add “CURRENTDISCONNECTDURATION = 200” on the last row of the [SIP Params] field.
  • Save the BOARD.ini
  • Send the “BOARD.ini” file from your computer to the device.
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