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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