Brekeke SIP Server Wiki

The call got cut off when the callee answers a call


Does codec match between UAs (phones)? Check if the codec setting is set correctly. 

If you use Brekeke PBX, check [RTP relay] settings at the ARS and on Options page.Please refer to “Brekeke PBX Administrator’s guide (Advanced)” for more details. 

If you are using Brekeke PBX, in particular situation, there are some cases that re-INVITE is sent out.If your phones do not support re-INVITE, please set [RTP relay] as ‘on (G.711u only)’, or ‘on’.

On a machine which Brekeke SIP Server is running, capture the UDP packets using a network analyzer, such as Ethereal, or Wireshark, and analyze the packets. If you need help for analyzing the packets, please contact our support.

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