Hello, Here's the scenario (no NAT):
SIP phone | | Opensips 1.5.1 & Mediaproxy 2.3.4 | | Asterisk w/ reinvites | | PSTN gateway engage_media_proxy() happens on all initial INVITEs. As many of us are too aware, Asterisk starts with relaying the media through itself. The canreinvite option allows it to send reinvites to the two endpoints, adjusting the connection information to allow the endpoints to send media directly between themselves. It seems as though these reinvites are confusing Mediaproxy in some cases. When an outbound call (SIP phone --> PSTN gateway) happens, all is well. Mediaproxy reports the caller_remote and callee_remote IP/ports correctly as the SIP phone and the PSTN gateway, respectively. An inbound call (PSTN gateway --> SIP phone) is less successful. After the reinvite, the SIP phone --> PSTN gateway audio is relayed correctly. But, the PSTN gateway --> SIP Phone audio still relays to Asterisk, not to the SIP phone. The caller_remote reports as Asterisk's IP/port, not as the PSTN gateway's. I see what's happening, but I don't know why. What's the next logical step in diagnosing this? Thanks, Jeff _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users