<div dir="ltr"><div>So, it's doing it because the cseq is out of order not because its a request pending.<br></div>That clarifies the reason for the 500 so that may alter OEJ's hypothesis.<div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Mon, Feb 3, 2014 at 5:44 PM, dotnetdub <span dir="ltr"><<a href="mailto:dotnetdub@gmail.com" target="_blank">dotnetdub@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Anthony - thanks - that is quite another level of logging! Wow..<br>
<br>
So the problem freeswitch is giving out about is this:<br>
<br>
nta.c:2967 agent_recv_request() nta: INVITE (102) going to existing leg<br>
nta.c:1341 set_timeout() nta: timer shortened to 200 ms<br>
nta.c:5441 incoming_callback() nta_leg(0xb6656790): out-of-order<br>
INVITE (102 < 103)<br>
<br>
It then sends the 500<br>
<br>
The CSEQ on the reinvite is 102 - the CSEQ on the UPDATE that asterisk<br>
sent just before was 103 ....<br>
<br>
Does this help or do you require more information.<br>
<br>
Many Thanks<br>
Brian<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
<br>
<br>
On 3 February 2014 22:50, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
> What you want is logs from FS side of things to try to catch more info.<br>
><br>
> sofia global siptrace on<br>
> sofia tracelevel alert<br>
> sofia loglevel all 9<br>
> console loglevel debug<br>
><br>
> And you have to catch the problem at the beginning of when it first starts<br>
> happening.<br>
><br>
><br>
><br>
><br>
><br>
><br>
> On Mon, Feb 3, 2014 at 1:08 PM, dotnetdub <<a href="mailto:dotnetdub@gmail.com">dotnetdub@gmail.com</a>> wrote:<br>
>><br>
>> Hi Mike,<br>
>><br>
>> Thanks for looking at this.<br>
>><br>
>> I have the log level at 7 and this is all that appears immediately<br>
>> before the invite and 500<br>
>><br>
>> [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/012345678<br>
>> [BREAK]<br>
>><br>
>> I can't see anything else relating to the reinvite.<br>
>><br>
>> Thanks<br>
>> Brian<br>
>><br>
>><br>
>> On 3 February 2014 16:20, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>
>> > Have you looked at the debug log from freeswitch to see why it is<br>
>> > sending the 500?<br>
>> ><br>
>> > On Feb 2, 2014, at 7:35 AM, dotnetdub <<a href="mailto:dotnetdub@gmail.com">dotnetdub@gmail.com</a>> wrote:<br>
>> ><br>
>> >> Hi All,<br>
>> >><br>
>> >> We are facing an issue where we have a customer using asterisk which<br>
>> >> insists on sending connected line updates via SIP UPDATES and<br>
>> >> REINVITES.<br>
>> >><br>
>> >> I have tried to mitigate this on the customer Asterisk by using all<br>
>> >> the config instructions you would expect to work but there are still<br>
>> >> some scenarios where asterisk insists on sending REINVITES with caller<br>
>> >> ID update to the trunk. I've also tried a number of things on the<br>
>> >> freeswitch side to try and persuade Asterisk not to send UPDATES or<br>
>> >> REINVITES without success...<br>
>> >><br>
>> >> SIP Dialog always starts off well but quickly descends into thousands<br>
>> >> upon thousands of UPDATES and INVITES<br>
>> >><br>
>> >> Freeswitch sends OK to every UPDATE and an OK to the first REINVITE<br>
>> >> but then starts sending 500 . Asterisk does ACK these errors but<br>
>> >> immediately sends another UPDATE (again FS sends ok) and INVITE which<br>
>> >> again we send back a 500 - you can see where I'm going here.<br>
>> >><br>
>> >> This can end up with 1000s of UPDATE,INVITE,500 going on in a SIP<br>
>> >> dialog and customer reports it is affecting the call audio while this<br>
>> >> is happening.. I still am unsure of why the audio is affected, the<br>
>> >> asterisk invites always have SDP so maybe asterisk is opening/closing<br>
>> >> the port very quickly or something...<br>
>> >><br>
>> >> I've attached a pastie of the first 48 packets of the SIP Trace. This<br>
>> >> particular call has 2067 SIP packets , mostly INVITE, 500<br>
>> >><br>
>> >> Olle (OEJ) comments on the Kamailio mailing list:<br>
>> >><br>
>> >> 'If freeswitch believes it already has an open INVITE transaction it<br>
>> >> should<br>
>> >> not respond with 500, it should respond with 491 request pending. In<br>
>> >> that<br>
>> >> case Asterisk will back off and retry.<br>
>> >><br>
>> >> Please check with the FreeSwitch people and file a bug report so that<br>
>> >> they<br>
>> >> can fix this issue. That's the long term solution, all the rest is<br>
>> >> just quick and<br>
>> >> dirty fixes. Seems like if this problem is still around, no one filed<br>
>> >> a bug report.'<br>
>> >><br>
>> >> Should I file a bug report or what are freeswitch users / dev thoughts<br>
>> >> on this?<br>
>> >><br>
>> >> Maybe there is something I can do in config or is this something that<br>
>> >> I must address with Asterisk... With older versions that some<br>
>> >> customers use I was able to stop reinvites completely. With version 11<br>
>> >> this functionality is still documented but doesn't work so well...<br>
>> >><br>
>> >> traces are <a href="http://pastebin.freeswitch.org/21924" target="_blank">http://pastebin.freeswitch.org/21924</a> truncated after 48<br>
>> >> packets.<br>
>> >><br>
>> >> Many Thanks,<br>
>> >> Brian<br>
>> ><br>
>> ><br>
>> ><br>
>> > _________________________________________________________________________<br>
>> > Professional FreeSWITCH Consulting Services:<br>
>> > <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>> > <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>> ><br>
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> > <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
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>><br>
>> _________________________________________________________________________<br>
>> Professional FreeSWITCH Consulting Services:<br>
>> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>><br>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>><br>
>> Official FreeSWITCH Sites<br>
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><br>
><br>
><br>
><br>
> --<br>
> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<br>
><br>
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><br>
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><br>
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><br>
<br>
_________________________________________________________________________<br>
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<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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<br>
Official FreeSWITCH Sites<br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div>
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