[OpenSIPS-Users] Unexpected loose_route() behavior

[Bogdan-Andrei Iancu]

Hi Matthew,

Could you post a SIP trace (from your opensips) of a such call - to see 
exactly what is wrong and what can be done ?

Regards,
Bogdan

Matthew Lehner wrote:
> Bogdan,
>
> Thanks, that makes sense now. The problem seems to be at my provider
> though. They rewrite the RURI to be my opensips and not the
> destination. Seems the provider is ignoring my lr=on and making it a
> strict route. I am still figuring out the difference between
> loose/strict. Would the best solution to this be to ignore any Route
> headers from my provider in this case?
>
> Matt
>
> On Thu, Jun 10, 2010 at 4:20 PM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro> wrote:
>   
>> Hi Matthew,
>>
>> you have a logical bug in your routing - all sequential requests are
>> routed in SIP based only on Route hdrs (via loose_route), so there is no
>> need to do manual routing for them. Manual routing is done only for
>> initial requests.
>>
>> Better see http://www.opensips.org/Resources/Webinars#toc11
>>
>> When the sequential request comes from the provider, it will have a
>> Route hdr pointing to your opensips and in RURI will be the IP of the
>> final Asterisk box (contact from 200 OK) -> after loose_route, opensips
>> will remove the Route hdr with its IP and the request will be routed
>> (RURI based) to proper Asterisk.
>>
>> Regards,
>> Bogdan
>>
>> Matthew Lehner wrote:
>>     
>>> I am setting up opensips to act as a proxy between a SIP trunk
>>> provider and more than one asterisk server. I am using alias_db to
>>> determine which asterisk server a particular DID/user should be
>>> relayed to. I am also using record_route() to ensure my proxy stays in
>>> the entire dialog of the call.
>>>
>>> The initial requests go through just fine, but subsequent requests in
>>> the same dialog from the SIP provider are not getting routed properly
>>> because of loose_route().
>>>
>>> When the request from the SIP provider arrives, it hits loose_route()
>>> and the RURI gets changed to sip:222.222.222.227;lr=on which does not
>>> contain a username and so alias_db can no longer match the call
>>> details and route the request to the proper asterisk server.
>>>
>>> The way I understood loose_route() was supposed to work is.. it checks
>>> the top-most Route header to see if it is the local proxy.. if it is
>>> it removes that Route and if there is another Route below it.. it will
>>> change the RURI to that.
>>>
>>> If I just don't do loose_route() on requests from the SIP provider,
>>> everything works as expected.. but this does not seem like the right
>>> solution to the problem.
>>>
>>> I have included debug output from opensips, along with some of my own logging.
>>>
>>> 333.333.333.x is the SIP provider
>>> 222.222.222.x is my opensips proxy
>>>
>>> Regards,
>>>
>>> Matt
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>       
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
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>>
>>     
>
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>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro