<div dir="ltr">What you want is logs from FS side of things to try to catch more info.<div><br></div><div>sofia global siptrace on</div><div>sofia tracelevel alert</div><div>sofia loglevel all 9</div><div>console loglevel debug</div>
<div><br></div><div>And you have to catch the problem at the beginning of when it first starts happening.</div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Mon, Feb 3, 2014 at 1:08 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 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 [BREAK]<br>
<br>
I can't see anything else relating to the reinvite.<br>
<br>
Thanks<br>
Brian<br>
<div class="HOEnZb"><div class="h5"><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 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 should<br>
>> not respond with 500, it should respond with 491 request pending. In that<br>
>> case Asterisk will back off and retry.<br>
>><br>
>> Please check with the FreeSwitch people and file a bug report so that 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 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 packets.<br>
>><br>
>> Many Thanks,<br>
>> Brian<br>
><br>
><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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