[asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

Michael Maier m1278468 at allmail.net
Wed Nov 30 11:42:39 CST 2016

Hello all!

I can see a strange problem during invite in dialog in the context of
timer handling.

Given is the following incoming call from provider at (2 at 2)
to my asterisk at (1 at 1):

After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the reinvite started by the provider as usual.

The expected reinvite by the provider is started during authentication
of the reinvite started by asterisk and is answered immediately by
asterisk with sip 481.

The answer of the provider after the resend of the reinvite came about
0.5s later and is sip 481, too.

=> The session obviously isn't known on both sides!

Asterisk therefore now drops the call (bye).

Does anybody has any idea about the reason why both members don't
recognize the existing session any more? I hope the attached sip trace
can shed some light on the problem.

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