List,
I have been using asterisk for a while now & finally ran into the problem that I hear so often... "one way audio".
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
0.0.0.0 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.
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.
Does anyone have an idea of what I'm doing wrong?
Here is what I'm using for my SIP entry:
[214]
type=friend
host=dynamic
context=blahblah
dtmfmode=rfc2833
restrictcid=no
insecure=yes
disallow=all
allow=ulaw
secret=mypassword
qualify=2000
nat=no
callerid="me" <214>
type=friend
host=dynamic
context=blahblah
dtmfmode=rfc2833
restrictcid=no
insecure=yes
disallow=all
allow=ulaw
secret=mypassword
qualify=2000
nat=no
callerid="me" <214>
Thanks,
bp
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