[Asterisk-Users] RE: Random disconnects

[Benjamin on Asterisk Mailing Lists]

On Sat, 2 Oct 2004 14:46:07 -0400, Shilliday, Jim
<jshilliday at ejcenter.org> wrote:
> My problem is random disconnects on both incoming and outgoing
> calls.  The phones are behind a firewall; the * box has a public IP.  I
> suspected that the disconnects were caused by * bridging the calls and
> then the connection through the firewall timing out in some way.  I
> tried setting reinvite=no on the SIP phones so that the audio would be
> routed through *, but it didn't seem to help.  Does anyone have a
> suggestion for a next step?

Yes, use sip debug on the Asterisk console to watch what's happening
shortly before disconnect. That should give you some clues where the
problem is.

And just in case you wondered, you can turn debugging off again by
entering "sip no debug".

rgds
benjk

> 
> Jim Shilliday
> IT Director
> Equal Justice Center
> 1315 Walnut St. Suite 400
> Philadelphia PA 19107
> 215-238-6970
> 
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