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@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@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@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
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