[OpenSIPS-Users] ACK doesnt get routed to endpoint
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Feb 28 11:38:19 CET 2012
Hi Arnold,
If I understand you right, instead on focusing on fixing the root error,
you are trying to cope with it in your script.
As I said, the ACK you receive on .90 (from the previous hops) is bogus
as it has incomplete route set (the contact info of callee is missing).
So, it is not a problem in your script, but a problem with the previous
SIP hops.
Also, to answer to your questions - yes the ACK is part of the same
dialog. But note that the dialog variables will be available (for ACK)
only after doing the loose_route() - here is where the ACK is matched
against the dialog. To check if the ACK matched the dialog (after
loose_route), check the $DLG_status variable - if it is NULL, your ACK
did not matched.
Regards,
Bogdan
On 02/28/2012 12:01 PM, Arnold Vriezekolk NETZOZEKER B.V. wrote:
> Thanks for the reply Bogdan,
>
> If the ACK has the same tag in the From header as the initial INVITE,
> isnt
> that considered the same dialog?
>
> What i'm trying to achieve is set an AVP variable that has some
> information
> about this call. Whenever i try to reach the variable at the point the
> ACK
> message comes in from our sip provider the variable is empty.
>
> How can i make sure that when i set a variable at the initial INVITE i
> can
> reach it when the ACK comes in? Script variables also dont seem to
> work for
> me because they get overwritten. I might be able to set a variable
> based on a
> unique descriptor based on the call.
>
> Best Regards,
>
> Arnold Vriezekolk
>
> On Fri, 24 Feb 2012, Bogdan-Andrei Iancu wrote:
>
>> Hi Arnold,
>>
>> The ACK you get in .90 is bogus, as it should have in RURI the .130
>> IP from the
>> 200 OK Contact. Actually the end-point info from 200 OK (the .130) is
>> not present
>> at all in the ACK, so basically the ACK cannot end up to that end
>> point at all.
>>
>> Regards,
>> Bogdan
>>
>> On 02/24/2012 04:47 PM, Arnold Vriezekolk NETZOZEKER B.V. wrote:
>> Hi,
>>
>> In our setup we have a connection to a sip provider for incoming
>> lines. We use
>> opensips with the uac_registrant module to connect to this sip
>> provider.
>>
>> Whenever a call comes in, i use rewriteuri() and t_relay() to
>> send it
>> to an
>> endpoint. When i pick up the call on my endpoint the ACK
>> message from
>> the sip
>> provider doesn't get routed back to my endpoint, but somehow gets
>> stuck in
>> opensips. Thus failing the call after 3 seconds.
>>
>> The ACK should be routed by the default sequential request
>> block in
>> the
>> request route[0] afaik but it doesnt.
>>
>> Can anyone point me as to where i can fix this problem?
>>
>> I attached a pcap dump of the call for debugging purposes.
>> Endpoint: sip:EEEEEEEE at netzozeker.tel.netzozeker.nl (95.97.29.130)
>> SIP provider: 89.184.172.54
>> OpenSIPS: 195.60.212.90
>>
>> If you need more information please let me know, i can attach the
>> opensips.cfg
>> and the log of opensips for more debugging purposes.
>>
>> Best Regards,
>>
>> Arnold Vriezekolk
>>
>>
>> _______________________________________________
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>> Users at lists.opensips.org
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>>
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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