We have an interesting issue:
One of our providers has two softswitches. Calls coming from the
first one are handled fine by asterisk, calls coming from the second
one and going through the first one are euhm... dropped half a
second into the RTP stream.
I have opened a ticket at Digium for it:
http://bugs.digium.com/view.php?id=10449
The output of "sip debug" is funny from line 366, where it is
transmitting and re-transmitting a lot of re-invites back to the
softswitch with CSeq 102.
Has somebody else seen this behaviour, and know how to resolve it?
Edwin
--
Edwin Groothuis | Personal website: http://www.mavetju.org
[EMAIL PROTECTED] | Weblog: http://www.mavetju.org/weblog/
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