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 > [EMAIL PROTECTED] > 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----------------- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
