Hello we did an experiment with SER, rtpproxy and Asterisk on 1 server (voip1). UA has sip proxy set with 'voip1', realm with 'voip2'. We also have 'domain' module installed. Asterisk runs at port 5061. Most of things works fine. We can do 'echo test' with Asterisk by rewriting host:port to 'voip1:5061'. However, when UA calls PSTN, we did the same thing to rewrite 'host, port' to 'voip1:5061'; then t_relay. It turns out Asterisk calls out correctly (the callee phone rings), but can't send response back to UA. The UA goes dead, and has to be reset. Asterisk shows an active 'sip channel' to this UA. It appears that Asterisk can't find the way back to SER. We can't figure out why 'echo test' works fine, but not real PSTN call. Can someone shed some lights on this issue? thanks! steven