<div>List,</div>
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<div>I have been using asterisk for a while now & finally ran into the problem that I hear so often... "one way audio". </div>
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<div>We were using our asterisk server with 1 NIC and a Public IP with multiple ATA's all in different locations with no problems at all. We have now decided to migrate our existing PBX to Asterisk. So I installed & configured a second NIC with a Private IP and configured a few ATA's on the same network with the NIC. The private NIC was configured with Gateway address of
<a href="http://0.0.0.0">0.0.0.0</a> so all the traffic would go out through to the public NIC instead of being rerouted through the NAT. All seems to work perfectly, I can make calls to the PSTN just fine (through one of our SIP carriers), but incoming calls from the PSTN (through a SIP carrier) have only 1 way audio.
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<div>When I call from one extension to another, it's fine... when I call from extension to pstn, it's fine... but PSTN to SIP device has 1 way audio. I tried turning off IPtables, but that made no difference. It still works fine when connecting to the public IP address instead of the private one.
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<div>Does anyone have an idea of what I'm doing wrong?</div>
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<div>Here is what I'm using for my SIP entry:</div>
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<div>[214]<br>type=friend<br>host=dynamic<br>context=blahblah<br>dtmfmode=rfc2833<br>restrictcid=no<br>insecure=yes<br>disallow=all<br>allow=ulaw<br>secret=mypassword<br>qualify=2000<br>nat=no<br>callerid="me" <214>
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<div>Thanks,</div>
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<div> bp</div>