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

[Clif Jones]

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. 
>>
>>
>>    
>>
>  
>