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. SpectraLink 8400 Wireless Telephone
      15. SAXA IP Netphone SX Phone
      16. Snom m3 VoIP wireless DECT phone
      17. Snom Phones
      18. Swissvoice IP10S Phone
      19. Ubiquiti UniFi VoIP Phones
      20. Vtech Business Phones
      21. Yealink SIP Phones
      22. 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. VoIP INNOVATIONS
    10. Voxbone
    11. VoIPUSER
    12. 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 TP-260

The TrunkPack® (TP)-260/SIP is a PCI VoIP communication gateway board from AudioCodes. 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 TP-260/SIP. For more information on this product, please go to http://www.audiocode.com

TP-260/SIP Hardware Installation

The TP-260 User’s Manual describes the installation and use of this unit. It can be found in the CD (“TrunkPack Digital Boards & VoP GW Series”) that shipped with the unit.

Using a Web Browser to Enable Service

The TP-260/SIP has an embedded web server which allows you to easily view and manage your gateway’s settings through a web browser. Please follow the instructions below carefully.

  1. Run a web browser application on the same network that the TP-260/SIP is connected.
  2. Log in to the setup menu by typing: http://device’s IP address.

The following is a guide for configuring the TP-260/SIP to work with Brekeke PBX to allow you to send and receive VoIP calls. Please note that only required parameters, within Brekeke PBX and SIP server domain, are discussed here. Most of them are self explanatory and there is no need to describe them all in details. You can always consult the User’s Manual of this product if you need in-depth technical information.

The document in this section is written based on “Mediant & TP Series SIP User’s Manual” by AudioCodes, version 4.6, Document#: LTRT-68803

A quick glance on the main menu bar, you will see 7 menus as shown below.

1. Quick Setup

Set up IP, SIP parameters, coder name, routing and trunk group tables.

2.Protocol Management – Protocol Definition – General Parameters

Configure the gateway’s general parameters

3. Protocol Management – Protocol Definition – Proxy & Registration

Configure the gateway’s proxy and registration parameters

4. Protocol Management – Protocol Definition – Coder

The current version of Brekeke PBX supports three codecs: g711Ulaw64k, g711Alaw64k and iLBC. The g711Ulaw64k codec is the most commonly used.

5. Protocol Management – Protocol Definition – DTMF & Dialing

Configure DTMF and dialing parameters

6. Protocol Management – Routing Tables – General Parameters

Configure general parameters for routing tables

7. Protocol Management – Routing Tables – Tel to IP Routing

Configure the gateway’s Tel -> IP routing tables. It’s used to route incoming telephone calls to IP addresses when proxy server is not used.

8. Protocol Management – Routing Tables – IP to Trunk Group Routing Table

This table is used to route incoming IP calls to trunk groups based on any combination of the destination phone prefix, source phone prefix and source IP address.

9. Protocol Management – Profile Definitions – Tel Profile Settings

You can define up to four different Tel Profiles. The gateway’s B-channel characteristic is defined based on each Tel Profile.

10. Protocol Management – Profile Definitions – IP Profile Settings

You can define up to four different IP Profiles. They are used in the Tel to IP and IP to Trunk Group Routing tables to associate different profiles to routing rules.

11. Protocol Management – Trunk Group

Use this table to assign trunk groups, profiles and logical telephone numbers to the gateway’s E1/T1 B-channels.

12. Protocol Management – Trunk Group Settings

This table is used to determine the method in which new calls are assigned to B-channels within each trunk group.

13. Advanced Configuration – Network Settings – IP Settings


14. Advanced Configuration – Channel Settings – Voice Settings

These menus allow you to set up the gateway’s channel parameters for voice, fax, RTP and IPmedia.

15. Advanced Configuration – Trunk Settings

This menu allows you to configure the parameters of a specific trunk. It’s very important that you set the T1 parameters in this menu to match with the settings of T1/E1 trunk.

After applying trunk settings, you should see the trunk status like this which indicates no alarm on trunk 1.

If you have encountered other alarm(s), please refer to the trunk status color indicator below to diagnose the problem before proceeding to the next section.

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: 
To=sip:([0-9]{7,25})@

Deploy Patterns: 
To=sip:$1@gateway_IP
------------------------
Making Test Calls

At this point, you are ready to make the test calls to ensure the T1/E1 trunk; TP-260/SIP and Brekeke PBX are in sync and working seamlessly.

  • Incoming calls: Call the telephone numbers of your T1/E1 voice circuit from your mobile or landline phone and make sure calls are connected to Brekeke PBX.
  • Outgoing Calls: Dial destination number from one of the extension of Brekeke PBX and make sure calls are connected to the destination number.
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