[asterisk-users] Outgoing SIP calls dropped after 30 seconds.
Kurt Knudsen
kurt.knudsen at gmail.com
Fri Nov 7 15:28:16 CST 2008
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check
back to see if it worked. Would be nice if it did :)
Thanks,
Kurt
On Fri, Nov 7, 2008 at 3:38 PM, Doug <Doug at natel.net> wrote:
> At 14:15 11/7/2008, SIP wrote:
> >Kurt Knudsen wrote:
> >> Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
> >> with a public IP address. We have our phone system setup as 172.16.2.x
> >> that connect through the SonicWall to Asterisk. Incoming calls work
> >> flawlessly and we no longer get one-way audio. We are only using SIP
> >> (3 trunks now, instead of 2) and having all 3 in use is not an issue.
>
>
>
> >> Question: Why does it sometimes work and sometimes not? This makes no
> >> sense and it happens on all phones. Any suggestions?
> >>
> >>
> >>
> >
> >
> >We see this on occasion. It sounds a lot like Asterisk doing its usual
> >routine of deciding that you can't POSSIBLY have a call going through
> >because it can't receive an ACK response properly. Asterisk tries
> >several times to send an ACK and get a response. If the remote system
> >routes ACKs differently than it routes everything else, often times
> >those ACKs get lost, and Asterisk assumes that the call can't be
> >working, so it destroys it.
> >
> >ACK handling is a bit tricky in the real world, and we've run across
> >countless incorrectly-configured SIP servers that don't handle it
> >properly, so calls to them last just about exactly 30 seconds and then
> >drop.
> >
> >There is, unfortunately, no way to turn off Asterisk's 'intelligent'
> >behaviour in this scenario short of possibly patching the code.
>
> http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html
>
>
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