Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues.
One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk, which then initiates unwanted Packet2Packet bridging. http://www.pastebin.ca/849961 <--- Working call. Line 280 shows a "SIP/2.0 183 Session Progress" and the RTP stream works as intended. I hung the call up before being answered in case you were wondering where the answer part of the debug occurs http://www.pastebin.ca/849965 <--- Non-working call. Line 235 shows a "SIP/2.0 180 Ringing", and for some odd reason, Packet2Packet bridging is initiated despite "canreinvite=no" being set. This in turn causes me some one way audio behaviour, and I've got no idea how to fix it. Anyone able to offer any suggestions. Regards, Cameron Jenkins _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
