[Freeswitch-users] Asterisk / Proxy / Freeswitch reinvites race condition
mike at jerris.com
Mon Feb 3 19:20:45 MSK 2014
Have you looked at the debug log from freeswitch to see why it is sending the 500?
On Feb 2, 2014, at 7:35 AM, dotnetdub <dotnetdub at gmail.com> wrote:
> Hi All,
> We are facing an issue where we have a customer using asterisk which
> insists on sending connected line updates via SIP UPDATES and
> I have tried to mitigate this on the customer Asterisk by using all
> the config instructions you would expect to work but there are still
> some scenarios where asterisk insists on sending REINVITES with caller
> ID update to the trunk. I've also tried a number of things on the
> freeswitch side to try and persuade Asterisk not to send UPDATES or
> REINVITES without success...
> SIP Dialog always starts off well but quickly descends into thousands
> upon thousands of UPDATES and INVITES
> Freeswitch sends OK to every UPDATE and an OK to the first REINVITE
> but then starts sending 500 . Asterisk does ACK these errors but
> immediately sends another UPDATE (again FS sends ok) and INVITE which
> again we send back a 500 - you can see where I'm going here.
> This can end up with 1000s of UPDATE,INVITE,500 going on in a SIP
> dialog and customer reports it is affecting the call audio while this
> is happening.. I still am unsure of why the audio is affected, the
> asterisk invites always have SDP so maybe asterisk is opening/closing
> the port very quickly or something...
> I've attached a pastie of the first 48 packets of the SIP Trace. This
> particular call has 2067 SIP packets , mostly INVITE, 500
> Olle (OEJ) comments on the Kamailio mailing list:
> 'If freeswitch believes it already has an open INVITE transaction it should
> not respond with 500, it should respond with 491 request pending. In that
> case Asterisk will back off and retry.
> Please check with the FreeSwitch people and file a bug report so that they
> can fix this issue. That's the long term solution, all the rest is
> just quick and
> dirty fixes. Seems like if this problem is still around, no one filed
> a bug report.'
> Should I file a bug report or what are freeswitch users / dev thoughts on this?
> Maybe there is something I can do in config or is this something that
> I must address with Asterisk... With older versions that some
> customers use I was able to stop reinvites completely. With version 11
> this functionality is still documented but doesn't work so well...
> traces are http://pastebin.freeswitch.org/21924 truncated after 48 packets.
> Many Thanks,
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