The calling part is a phone directly connected to a Voip GW. The Voip GW is in a LAN having private IP. The call is sent to an AS5xxx series. It is authenticated by the public IP of that LAN, the call is connected and there is one way voice.
On As5xxx I see the call connected (with "show call active voice brief" command), but the IP from where the call is coming is the private IP of the Voip GW, and not the public IP of the LAN where the GW is. So - i think the cisco don't know where to send the RTP packets, and the voice back is lost.
There is a command on cisco for this h323 calls issue (like "nat symmetric check-media-src" for sip calls) ?
Maybe you could use the h323-gateway voip bind srcaddr ip_address command, I don't think I have tried it on an analog connected port, but it might work. Tie it to an address the far end can route to, you can use a loopback interface for this too.
Also, I had kept some notes on H323 and NAT from a while ago, you can also have a pix handle it if it is version 6.3 or above with the fixup protocol h323, then it will translate the voice traffic address for you.
These are the paths to get to each CCX logs through CLI. They may be helpful if you are having issues accessing RTMT or downloading logs through it.
If you want to download them you have to prefix "file get " and you can add one of the options (re...