Hi List,
Someone may have run into this problem. Very strange.
I have a customer running 1.422. They use a digium ISDN card connected to an
primary rate for their inbound currently.
We have tested inbound SIP from one of our trunks. We use these trunks with
all our asterisk customers without an issue.
With this Asterisk box when we answer an inbound SIP call to an extension
literally after .5 seconds (500ms) the audio just dies going from the
extension to the callee... The extension call still hear the caller.
If we point the DID at a conference audio works perfectly. If we point it to
an IVR which then points to the extension the audio is perfect both
directions.
The SIP traces look perfect, identical SDP if going to an extension or a
IVR.
Any clues?
How would I go about debugging this, the CLI output looks fine..
Asterisk 1.4.22 built by root @ asterisk on a i686 running Linux on
2010-05-13 16:19:20 UTC
Thanks in advance.
Brian.
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