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 Medicatrix 2102

The Mediatrix 2102 is a high-quality and cost efficient VoIP gateway connecting SOHOs to an IP network, while preserving investment in analog telephones and faxes. It connects up to two analog phones and/or faxes, as well as a PC or a home router to an IP network over a single broadband connection. www.mediatrix.com

Configure Mediatrix 2102

Once the IP address is configured you can access Mediatrix 2102 FXS Gateway webpage by http://192.168.10.1/ (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 and Restart

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

Open the Unit Manager Network software which you can obtain from Mediatrix web site and discover the device.

“SIP Registrar” :192.168.0.167 (Ex)
“SIP Proxy”:192.168.0.167(ex)
“User Name”/Phone number: 222/333 (Ex)

Click on [OK] button

2. Networking Setting

2.1 System > WAN

“WAN IP Address”: 192.168.0.99(Ex)
“WAN Network Mask”:255.255.255.0 (Ex)
“Default Gateway”:192.168.0.2 (Ex)
“Primary DNS”:XXX.XXX.XXX.XXX
“Secondary DNS”:XXX.XXX.XXX.XXX

Click on [Submit] button

 

Brekeke SIP Server’s Registration Page

Click the Registered tab of Brekeke SIP Server admintool.

Mediatrix 2102 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 *.

  1. Entry the “Digit Map Allowed” 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.
To enable DTMF Transport Using SIP INFO by Unit Manager Network

  • In the voiceIfMIB, set the DTMF transport type in the voiceIfDtmfTransport variable (voiceIfDtmfTransportTablegroup): outOfBandUsingSignalingProtocol
  • In the sipInteropMIBsipInteropDtmfTransportBySipProtocol, set the DTMF transport type in the sipInteropDtmfTransportMethod variable : infoDtmfRelay and in the sipInteropDtmfTransportDuration variable:160
  • Set the DTMF duration sent in the INFO message when using the infoDtmfRelay method to transmit DTMFs in thesipInteropDtmfTransportDuration 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 analogScnGwInterDigitDialDelay variable. This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone number. This is useful when the Mediatrix 2102 receives DTMFs out-of-band faster than it can signal them.
  • Restart the Mediatrix 2102 so that the changes may take effect.
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