Yes, you need to further isolate the problem using process of elimination. The message tells you that Asterisk didn’t receive media for a period of time and thus disconnected the call, so using that basis you isolate it further by ruling out segments to further isolate things and rule things out.
If you are “getting complaints” then likely you will know who is complaining at what time and what endpoints were involved, given those data points
will save you a huge amount of time isolating the exact point when the RTP disappeared. It wont tell you why but you will likely see dropped packets, perhaps out of order, perhaps delayed packets, this can help further diagnosing network problems.
Thanks Dicko. Thanks for suggesting pcapsipdump. Downloaded it and will try it out. Also in SIP SETTINGS the RTP timeout is set to 30. If I change it to 60 would that help in the meantime? I spoke to the person at the extension with the dropped calls for more than 10 minutes. That call did not drop.
It would alter the time that no audio is required before disconnecting the call. If the drop is temporary and below 60 seconds then it would not disconnect. Whether that would help depends on what exactly is happening.
If pcapsipdump is run as:
pcapsipdump -v 2 -i eth0 -d /var/tmp/ -R rtp port 5060 or portrange 10000-20000
it fails with:
Capturing on interface: port
Couldn’t get netmask for interface port: SIOCGIFADDR: port: No such device
Couldn’t open interface ‘port’: port: No such device exists (SIOCGIFHWADDR: No such device)
Tried moving the port 5060 part closer to command pcapsipdump but that doesn’t help.