That SIP message is actually OpenSIPS receiving the BYE from the customer, the problem is OpenSIPS is not forwarding/relaying that BYE message to the Asterisk server that is handling the call, so Asterisk doesn't know the call ended, and it's still letting the phone ring (and possibly be answered) on the other side of the call. I will get a more complete SIP trace up.
My environment basically looks like this: Customers -> OpenSIPS Load Balancing -> Asterisk Pool -> OpenSIPS -> Vendors Russ -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Anca Vamanu Sent: Friday, December 10, 2010 5:08 AM To: [email protected] Subject: Re: [OpenSIPS-Users] Handing BYE instead of CANCEL before Answer Hi Russell, I see from the trace that the BYE is forwarded by OpenSIPS to the other end here: E..:....@[email protected] .B.O}j......& .BYE sip:[email protected] SIP/2.0 Max-Forwards: 67 To: 11110702923002334053 <sip:[email protected]>;tag=as6fbb0b2b From: <sip:[email protected]>;tag=3500915879-587280 Contact: <sip:customer.address:5060;transport=udp> Call-ID: [email protected] CSeq: 2 BYE Via: SIP/2.0/UDP public.address;branch=z9hG4bK9188.2b1bd8b4.0 Via: SIP/2.0/UDP customer.address:5060;rport=5060;received=customer.address;branch=z9hG4bK15b3b0b1ea8e4d40f85f0cc6056f0968 Content-Length: 0 But I don't know of a way you could replace that Bye with a Cancel.. Regards, -- Anca Vamanu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
