We use asterisk (V 13.0.1) on (CentOS 6.7) for voice recording.
The calls, which should be recorded come from SIP trunk from another PBX. Recording works, except one problem. Sometimes the recorded wav files is not complete:
Example:
Voice: Test - One - Two - Three - Four - Five
WAV-File: Test - One - Four - Five
I made a TCP dump and found the complete voice in RTP packages, but one thing is noticeable:
When the wav-File is imcomplete, there is a change in RTP source. This seems, there is a problem in recording, when the RTP source changes in the middle of the call.
Example (from TCP-Dump):
RTP from IP address 1 contains: Test - One - Two - Three -
RTP from IP address 2 contains: Four - Five
The result is a wav file containing the first part from ip 1 ("Two - Three - " is missing) and the complete stream from ip 2.
Does anyone know there is an issue when changing RTP source during the call?