[OpenSIPS-Users] OpenSIPS swallows BYEs

[Bogdan-Andrei Iancu]

Hi David,

ok, so the BYE is sent to the wrong destination - how do you route BYE 
in your script? do you use the standard "loose_route" block ?

Regards,
Bogdan

David Santiago wrote:
> As I said it is being forwarded to the jain slee server that
> originated it. The effect is the same as if it was replied.
>
> The flow is:
> step 1:   [JAIN SLEE SERVER]   ---BYE-->    [OPENSIPS]        [VOIP PROVIDER]
> step 2:   [JAIN SLEE SERVER]   <---BYE--    [OPENSIPS]        [VOIP PROVIDER]
>
> that BYE should be forwarded to the VOIP provider, instead.
>
> I'm having a look at how to configure the b2bua thing should this
> problem be fixed.
>
> Thanks for the follow up, Bogdan.
>
>
> David
>
> On Fri, Oct 8, 2010 at 1:38 PM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro> wrote:
>   
>> Hi David,
>>
>> my question is what happens with the BYE:
>>   - is it replied by opensips ?
>>   - is it forwarded to some whatever destination ?
>>
>> Regards,
>> Bogdan
>>
>> David Santiago wrote:
>>     
>>> Bogdan, right now it's being forwarded again to the slee server who sent
>>> it, as I'm basically using the configuration provided in
>>> http://www.opensips.org/html/docs/modules/1.6.x/dispatcher.html
>>>
>>> On Wed, Oct 6, 2010 at 4:16 PM, Bogdan-Andrei Iancu
>>> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>>>
>>>    Hi David,
>>>
>>>    Is the BYE replied or forwarded to whatever destination ?
>>>
>>>    probably  your record routing is somehow broken. OpenSIPS may
>>>    misroute the BYE because the invalid route set - posting the 200
>>>    OK for INVITE and the BYE will help in investigating this.
>>>
>>>    Regards,
>>>    Bogdan
>>>
>>>    David Santiago wrote:
>>>
>>>        Hi all,
>>>
>>>        I have a running OpenSIPS installation that I'm using for
>>>        testing purposes.
>>>
>>>        The fact is that I'm forwarding requests from a voip provider
>>>        to a jain slee server and everything is working fine (INVITEs,
>>>        ACKs, RTP flow,...), except for the BYEs generated from the
>>>        server side. They reach the OpenSIPs proxy and are not
>>>        forwarded to the voip provider in order to finish the call.
>>>
>>>        I'm not sure if I have to manually setup a route for this to
>>>        happen, or if this behaviour is only available by using the
>>>        B2BUA approach in OpenSIPS.
>>>
>>>
>>>        Thanks a lot!
>>>
>>>        David
>>>
>>>  ------------------------------------------------------------------------
>>>
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>>>
>>>
>>>
>>>    --     Bogdan-Andrei Iancu
>>>    OpenSIPS Bootcamp
>>>    15 - 19 November 2010, Edison, New Jersey, USA
>>>    www.voice-system.ro <http://www.voice-system.ro>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>       
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 15 - 19 November 2010, Edison, New Jersey, USA
>> www.voice-system.ro
>>
>>
>>     
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro