Greetings.
I am working on setting up an asterisk server (SIP only) and am running into
a few issues getting RTP working correctly.
Here is our setup:
SIP Client (Public IP) <---> Asterisk Server (Public IP/Private IP)
<-->
Nortel CSG (Internal IP) <--> PSTN
So far we have SIP to SIP working through Asterisk without any problems
(using various sip clients).
When I call from the PSTN to the CSG, here is what I see in the asterisk
console:
-- Executing Dial("SIP/10.10.100.40:5060",
"SIP/mgamble") in new stack
-- Called mgamble
-- SIP/mgamble-7fdd is ringing
-- SIP/mgamble-7fdd answered SIP/10.10.100.40:5060
-- Attempting native bridge of SIP/10.10.100.40:5060 and
SIP/mgamble-7fdd
The SIP/mgamble extention rings, however, when I pick up the phone I get no
audio in ether direction. Is there someway to better debug the 'native
bridge'?
Going the other way (from SIP to the PSTN) I can hear the audio from the SIP
client over the PSTN, but I can't hear the PSTN audio comming back to the
SIP client.
Is anyone running a private SIP gateway behind asterisk like in this
seniario? What needs to be done to get audio going both ways? Any hints?
Thanks in advance,
M. Gamble
