Hello, I'm using B2B to play a soundfile before connecting to an endpoint. My opensips is connected to a SIP provider for incoming calls.
Whenever a call comes in from my provider the b2b will send an invite to asterisk to play a soundfile. Whenever the soundfile is done playing the b2b will invite the endpoint. At packet 18 in the pcap dump[1] i hung up the call from the provider side, but the provider keeps sending the "200 OK" to establish a dialog. Without the ACK being sent from the opensips (or endpoint) side there will never be a dialog. According to the RFC3261[2] a BYE must not be send while there hasn't been an ACK to establish the dialog. What happens now is that the caller hangs up the call and the callee is still ringing. The callee will ring for 30 seconds or so before the provider times out the call and sends a BYE. If the callee would pick up his ringing phone, nobody would be on the other end because the caller had hung up the phone. So my question is: how do i solve this problem. Should my provider send a CANCEL when the caller hangs up the call? Should OpenSIPS send the ACK to establish the dialog? Thanks in advance, Best Regards, Arnold Vriezekolk [1] Pcap Dump: http://vriezekolk.org/~tuxx/03-07-b2b.pcap [2] RFC3261: However, the callee's UA MUST NOT send a BYE on a confirmed dialog until it has received an ACK for its 2xx response or until the server transaction times out. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
