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

[dotnetdub]

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
>
>
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