> Second thing I'd look at is the snoms. If you can grab the logs off one
> just after this happens I feel it would point you in the right direction.
> IIRC they also have some kind of packet dump facility? If this is
> happening 10% of the time it might be worth setting this up on one to see
> what really is happening.
Today we have been capturing traffic from both ends, on the Snom and on
What we are finding is that for the broken calls the RTP traffic only
gets set up in one direction (phone to asterisk) yet audio never makes
it back to the calling party.
It seems like the sip call to the agent is only being partially setup.
recovering the audio from the tcpdump shows that when this happens the
audio going from the phone to the asterisk server contains the agent
We are not seeing anything that looks untoward in how the sip call is
being setup, but we may still be looking in the wrong places.
Bouncing the call back to Queue does not connect the broken call back to
the Music on hold.
Holding and unholding the call does not reestablish a working session
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