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

[Anthony Minessale]

What you want is logs from FS side of things to try to catch more info.

sofia global siptrace on
sofia tracelevel alert
sofia loglevel all 9
console loglevel debug

And you have to catch the problem at the beginning of when it first starts
happening.






On Mon, Feb 3, 2014 at 1:08 PM, dotnetdub <dotnetdub at gmail.com> wrote:

> Hi Mike,
>
> Thanks for looking at this.
>
> I have the log level at 7 and this is all that appears immediately
> before the invite and 500
>
> [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/012345678
> [BREAK]
>
> I can't see anything else relating to the reinvite.
>
> Thanks
> Brian
>
>
> On 3 February 2014 16:20, Michael Jerris <mike at jerris.com> wrote:
> > 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
> >> REINVITES.
> >>
> >> 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,
> >> Brian
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140203/5291f868/attachment.html