Fw: [Asterisk-Users] Possible Sip logic bug?

[Rich Adamson]

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. 
> >>
> >>
> >>    
> >>
> >  
> >
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