[asterisk-users] audio trouble with asterisk, help very much appreciated

Jody Gugelhupf knueffle at yahoo.com
Thu Nov 29 19:07:33 CST 2012

 From: Jody Gugelhupf <knueffle at yahoo.com>
To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> 
Sent: Thursday, November 29, 2012 10:23:01 PM
Subject: audio trouble with asterisk, help very much appreciated

Hi there :)

first about my setup, running centos 6.2, asterisk, freepbx 
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 Asterisk and freepbx are on, my extension I make the calls from is '903' and is on
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. 
When I make an outbound call, the incoming audio works without flaws, however my outgoing audio drops after a minute or so. 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: yes (but have also tried 'no', 'never', and 'route' whilst keeping same settings below with same audio problems results as described above)
IP Configuration: Static IP
External IP:
Local Networks:
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:
Would appreciate any help. Thank you in advance! :)

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