Sorry Clif, as a professional working with protocol analysis at corporations in more than 40 states, I should have known better. Never gave it a thought the issue could have been earlier in the call/session setup. I'll dig into that, and if still need help/suggestions will post the full debug trace.
Rich ------------------------------------- > It is very important (at least to me) to have the whole SIP call flow. > That is, I must see the initial > INVITE come from the originating phone all the way to the last message. > I can only speculate at > this point but it appears that the second leg (destination) may never > have ACK'd the call which > could have Asterisk in a bad state. I cannot be sure of this without > the entire flow but if this is the > case, not only do you have a config problem, Asterisk has an unhandled > error state. Did you > answer the destination? Did it have 2-way voice path? > > Rich Adamson wrote: > > >Clif and all... > > > >At the bottom of this post is the "sip show debug" for the problem. > >The underlying problem (again): when C7960 hangs up on working conversation, > >the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway. > > > >Any suggestions would be greatly appreciated. > > > >Rich > > > > > > > >>Try it again after executing: "sip debug" and give us the actual SIP > >>messages. The devil > >>is usually in the details. > >> > >> > >> > >> > > > > > _______________________________________________ > 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
