Brekeke PBX Wiki

One-way call, and no voice at all

1. Please check the phone’s volume (a microphone, a headset, or a speaker).


2. When you use a SIP client which implements SIP protocol improperly, SIP session fails and may get disconnected after about 1 minutes.


3. When sounds do not go through properly, there are cases that either the UA (phone) or Brekeke SIP Server disconnect the session. Please check timeout setting.

  • Brekeke SIP Server : Go to [Configuration] – [RTP] – [RTP Session Timeout (ms)]
  • Brekeke PBX : Go to [Options] – [Media Server system settings] – [RTP Session Timeout (ms)]


4. Since ITSP tends to use old version of SIP standard, there are some cases that they can not process lr parameters. Please set [B2B-UA mode] on from SIP Server >[Configuration]>[SIP], or set $b2bua = true in the applicable deploy pattern in the dial plan.


5. Does both UAs (phones) codec match? Check if the codec setting is set correctly.
For Brekeke PBX, check RTP relay setting. If [RTP relay] setting is selected to “on (G.711u only)”, or “off (G711u only)”, please check if the SIP client supports G.711ulaw. Otherwise, please set it to “on (G711uonly)”. Some ITSPs or SIP client do not work unless you set it to “on (G711u only)”.
For more details about RTP relay, please refer to ”Brekeke PBX Administrator’s guide (Advanced)”.


6. When you use Brekeke PBX and neither of your machine or ITSP supports re-INVITE, you need to set [RTP relay] to “on (G711u only)” or “on”. For more details about RTP relay, please refer to ”Brekeke PBX Administrator’s guide (Advanced)”.


7. When NAT traversal is use (when you have communication over mutual networks), please refer “Brekeke SIP Server Administrator’s guide – NAT traversal” for more details.
At clients side router, RTP ports forwarding may be necessary.


8. For Brekeke products behind NAT, please check the network settings at Brekeke SIP Server. There is a chance that ports for sound communication are not open by a router or a Firewall.
Check wiki posts about port forwarding and firewall settings.


9. On a machine which Brekeke SIP Server is running, capture the UDP packet using a network analyzer, such as Wireshark, and contact our support team.


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