[OpenSIPS-Users] OpenSIPS swallows BYEs

[David Santiago]

I'm not explicitly routing that BYE, but as Andrew Pogrebennyk pointed
out, since I'm "rewriting the Contact header with OpenSIPS address it
is expected that BYE won't go any further than OpenSIPs proxy" and
therefore my only solutions seems to be the configuration of the b2bua
modules. Isn't it?

On Mon, Oct 11, 2010 at 6:35 PM, Bogdan-Andrei Iancu
<bogdan at voice-system.ro> wrote:
> 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
>>>>
>>>>
>>>>  ------------------------------------------------------------------------
>>>>
>>>>       _______________________________________________
>>>>       Users mailing list
>>>>       Users at lists.opensips.org <mailto: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 <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
>
>