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 Mediatrix 1104(FXS Gateway)

This document explains how to use Brekeke SIP Server with Mediatrix 1104(FXS Gateway). The Mediatrix 1104 is a high-quality and cost efficient VoIP gateways connecting small offices to an IP network. The Mediatrix 1104 is a VoIP access device equipped with four FXS ports, one 10/100 BaseT Ethernet port and one PSTN bypass port. It connects analog phones or fax machines and legacy PBX and Key Systems to an IP telephony network.

Configure Mediatrix 1104

Once the IP address is configured you can access Mediatrix 1104 FXS Gateway webpage by (example of selected IP address).Please follow the following steps once the webpage opened.

  1. Enter User name: admin (default)
  2. Enter Password: 1234 (default)
  3. Click on [OK] button
SIP Setting

The FXS 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)
“Registrar Host “ : (Ex)
“Registrar Port “: 5060 (Ex)
"Proxy Host” : (Ex)
“Proxy Port”: 5060 (Ex)
“Unregistered Port Behavior”: enable port
“Port User Name”: 222(Ex)
Click on [Submit] button
2. SIP Authentication Setting: SIP > Authentication

Entry the “Index Realm”:
Entry the “Username”: 222 (Ex)
Entry the “Password”:1234 (Ex)
Click on [Submit] button
Brekeke SIP Server’s Registration Page

Click the Registered tab of Brekeke SIP Server admintool.

Mediatrix 1104 is registered with Brekeke SIP Server.

Additional Telephony Setting
1. Call Pickup: Telephony > Digit Maps

Call Pickup is a function of Brekeke PBX that allows users to dial “*” to answer incoming calls from any extension in the same Call Pickup group. By default, * is not allowed to dial. Please complete the following setting to allow asterisk *.

  1. Entry the “Digit Map” in the first row: *.T
  2. Digit Map Characters:
    T X . *
    The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished dialing and the SIP Server can make the call Matches any digit, excluding “#” and “*” Indicates a choice of matching expressions (OR). use “ *” to answer incoming calls from any extension in the same Call Pickup group
2. DTMF Setting: Telephony > CODEC

Set DTMF Transport Using SIP INFO, if RTP relay = off in Brekeke PBX. The following explains how to set Mediatrix to use SIP INFO.

  1. Set “DTMF Transport”: outOfBandUsingSignalingProtocol
  2. To enable DTMF Transport Using SIP INFO by Unit Manager Network
    • In the voiceIfMIB, set the DTMF transport type in thevoiceIfDtmfTransport variable (voiceIfDtmfTransportTable group):outOfBandUsingSignalingProtocol
    • In the sipInteropMIB> sipInteropDtmfTransportBySipProtocol, set the DTMF transport type in the sipInteropDtmfTransportMethodvariable : infoDtmfRelay and in thesipInteropDtmfTransportDuration variable:160
    • Set the DTMF duration sent in the INFO message when using the infoDtmfRelay method to transmit DTMFs in the sipInteropDtmfTransportDuration variable. This value is expressed in milliseconds (ms). The default value is 100 ms
    • In the analogScnGwMIB, set the DTMF duration when using the infoDtmfRelay method to receive. DTMFs in the analogScnGwDtmfDuration variable. This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone number.
    • Set an inter-digit dial delay in the analogScnGwInterDigitDial Delay variable. This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone number. This is useful when the Mediatrix 1104 receives DTMFs out-of-band faster than it can signal them.
    • Restart the Mediatrix 1104 so that the changes may take effect.
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