> That worked like a champ. Now I'm facing another issue: > > When a sip phone terminates the call, asterisk sends the BYE to the > wrong SIP URI... Instead of sending it to [EMAIL PROTECTED] > it sends it to [EMAIL PROTECTED] > > Unless I'm misuderstanding the rfc asterisk is not properly > handling the CONTACT header.
Sorry I left out some very important details in this email. The end result is that asterisk is sending the bye to the wrong sip URI and receiving a 404 response. This causes asterisk to keep the RTP channel open even after the call is supposed to be torn down as seen here: carbon*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 172.20.16.7 (None) 736d2dcb343 00101/201153740 00000ms 0000ms UNKN 172.20.16.7 (None) 4c98f793676 00101/1293907020 00000ms 0000ms UNKN 172.20.16.7 (None) 4be294900ba 00101/1744544499 00000ms 0000ms UNKN 3 active SIP channel(s) The PSTN Gateway also keeps the connection active as it hasn't received a BYE request. Thanks again for all your help -z _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
