[asterisk-users] Outgoing SIP calls dropped after 30 seconds.

[Kurt Knudsen]

That seems to have sort of worked. It seems the phone decided to end
the call this time, instead of Asterisk and now the call is dangling
inside of 'sip show channels'.

So that solution didn't work :(

On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen <kurt.knudsen at gmail.com> wrote:
> 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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>