Anyone recognize this as a sip logic bug? Example Case: C7960 -> * -> sip gateway -> pstn (sip gateway config'ed with canreinvite=no, but shouldn't have an impact on this.) Outgoing call initiated from C7960. Call is completed and conversation is very much normal. All equipment on the same wire; no nat. The C7960 user hangs up the phone. Pkt flows (as observed by sniffer) are: C7960 sends sip BYE packet to * * returns 200 OK * sends INVITE to sip gateway <<================ where is BYE? sip gateway responds with 100 Trying sip gateway responds with 200 OK sip gateway responds with 200 OK sip gateway responds with 200 OK The end result, the sip gateway does not drop the pstn line until the "called" number hangs up. It would appear that asterisk has an issue dropping the call. When the C7960 issues the BYE, I would expect * to send a BYE to the sip g/w. Is this a * logic problem (or my logic problem)? (I'm actually running CVS-12/04/03-14:24:40 and has been very stable in this production environment. Is it time to update this one even though it is 99% sip hardphone based?) Rich
Anyone have comments on this? Really could use some suggestions or ideas why this is happening. Thanks. Rich ------------------------> Anyone recognize this as a sip logic bug? > > Example Case: > C7960 -> * -> sip gateway -> pstn > (sip gateway config'ed with canreinvite=no, but shouldn't have an > impact on this.) > > Outgoing call initiated from C7960. Call is completed and conversation > is very much normal. All equipment on the same wire; no nat. > > The C7960 user hangs up the phone. Pkt flows (as observed by sniffer) > are: > > C7960 sends sip BYE packet to * > * returns 200 OK > * sends INVITE to sip gateway <<================ where is BYE? > sip gateway responds with 100 Trying > sip gateway responds with 200 OK > sip gateway responds with 200 OK > sip gateway responds with 200 OK > > The end result, the sip gateway does not drop the pstn line until the > "called" number hangs up. > > It would appear that asterisk has an issue dropping the call. When the > C7960 issues the BYE, I would expect * to send a BYE to the sip g/w. > Is this a * logic problem (or my logic problem)? > > (I'm actually running CVS-12/04/03-14:24:40 and has been very stable > in this production environment. Is it time to update this one even > though it is 99% sip hardphone based?) > > Rich > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users---------------End of Original Message-----------------