[asterisk-dev] Intermittent one-way audio and call failure on trunk

[Taylor, Jonn]

First off i open a bug on this a few weeks ago and was told that it was 
a network config problem but rolling back to a previous revision trunk 
makes the problem go away.

Have running test system at home on my gateway server that is running 
CentOS 5 i386, dual nic's, using freepbx 2.10. Using SIP, Unistim and 
IAX devices. SIP trunk provider is bandwidth.com(level3).

Current version of trunk I am getting 2 problem. Sometimes the phones do 
a partial ring and hangup and the second is the call will ring you can 
answer the call but get one-way audio, caller can not hear you. If you 
put the call on-hold sometimes you can get the audio to work.

Here are the two files with the SIP debug turned on.

http://www.taylortelephone.com/Files/failed-call.txt

http://www.taylortelephone.com/Files/one-way-audio.txt

Jonn