[Asterisk-Users] RTP / SIP routing issues

[Matthew M. Gamble]

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