[Freeswitch-users] Asterisk / Proxy / Freeswitch reinvites race condition

dotnetdub dotnetdub at gmail.com
Sun Feb 2 15:35:18 MSK 2014

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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