first about my setup, running centos 6.2, asterisk 18.104.22.168, freepbx 22.214.171.124.
I have a modem/router with NAT enabled. Asterisk and my extension from which I make the call are on the same local network, behind the modem/router.
I have forwarded ports 10000-20000 to asterisk and configured asterisk accordingly.
My external IP is 126.96.36.199. Asterisk and freepbx are on 192.168.2.199, my extension I make the calls from is ‘903’ and is on 192.168.2.202.
I have configured my extension and connected it. Also I have setup sip trunks and configured outbound rules etc.
This all works fine. When I receive calls, all works great, I have two way audio without any trouble during the whole call.
When I make an outbound call, the incoming audio works without flaws, all the time, however my outgoing audio drops after a minute or so, sometimes after 5 minutes. So first the other person can hear me then not. On some calls the outgoing audio starts working again after a bit, but then drops again.
In freepbx in the ‘asterisk sip settings’ I’m not sure how to set the NAT settings properly. Currently:
NAT: no (but have also tried ‘yes’, ‘never’, and ‘route’ whilst keeping same settings below with same audio problems results as described above)
IP Configuration: Static IP
External IP: 188.8.131.52
Local Networks: 192.168.2.0/255.255.255.0
Would this be correct?
I have run ‘sip set debug on’ and ‘rtp set debug on’ to see what happens during the call, the output is here:
I had some help with interpreting the logs on asterisk forum (http://forums.digium.com/viewtopic.php?f=1&t=85004&p=181098&sid=e97645ecfbe99ff6c1cc394a283d2393#p181098) basically as I was told, there are no re-invites, so I changed in freepbx, ‘asterisk sip settings’ the ‘re-invite behaviour’ setting to ‘yes’, no effect though. The calls are made to mobile phones and/or pstn landlines that are not connected to asterisks. They are not the problem (suggested in post). I got no clue any more what I can do.
Would appreciate any help. Thank you in advance!