[OpenSIPS-Users] 404 Not Here

[Bogdan-Andrei Iancu]

Well, escalating the problem will be the right thing to do.

As a workaround on your side, you could try to enable the topo-hiding on 
the dialog module, for your calls - this will take care of the contact 
issue.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 02/26/2013 04:00 PM, brad smith wrote:
> Bogdan,
> Thanks for responding.
> I am using vitelity for my upstream; I will send them a ticket.  If 
> they fail to act, do you have any suggestions...switch carriers? any 
> config change?
>
> Thanks again,
> Brad
>
>
> On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Brad,
>
>     Thinks are a bit more complicated, it seems....
>
>     In the INVITE your opensips sends to 64.....93 IP, you have the
>     Contact with 192.168.1.21 (priv IP of asterisk).
>
>     When you receive the BYE from 64.....93 IP, the Route hdrs are ok
>     (the 2 hdrs added by opensips to reflect the interface exchange),
>     but the RURI is wrong - it must be the contact from the INVITE you
>     sent, but it seems to be the IP of your opensips - this makes
>     opensips to do act as strict router and not like a loose
>     router....and routing gets broken.
>
>     So, the 64.....93 party or some other behind it, screw up the
>     Contact in the your INVITE and this alters the in-dialog requests
>     - you should check with the upstream guys.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>
>     On 02/25/2013 04:36 PM, brad smith wrote:
>>     I just tested an outbound call (Asterisk originate) without
>>     bridging and get the same '404 not here' if that helps.
>>
>>     Thanks again,
>>     Brad
>>
>>
>>     On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu <vladpaiu at opensips.org
>>     <mailto:vladpaiu at opensips.org>> wrote:
>>
>>         Hello,
>>
>>         Seems the incoming BYE does not have any Route headers, and
>>         the loose_route() function returns false.
>>
>>         Since you have dialog support in your script, try
>>
>>         if (has_totag()) {
>>         # sequential request withing a dialog should
>>         # take the path determined by record-routing
>>         if (loose_route() || match_dialog()) {
>>
>>         This way you will force matching of dialog sequential
>>         requests that have no Route headers.
>>
>>         Best Regards,
>>
>>         Vlad Paiu
>>         OpenSIPS Developer
>>         http://www.opensips-solutions.com  
>>
>>
>>         On 02/24/2013 02:57 AM, brad smith wrote:
>>>         Hello,
>>>
>>>
>>>
>>>         I am currently running opensips 1.8.1 no tls. It
>>>         is
>>>
>>>         multi-homed with a public and private address.
>>>         I have a asterisk
>>>
>>>         1.8.19 in the lan that is connected to
>>>         opensips via lan
>>>
>>>         address.
>>>
>>>
>>>
>>>
>>>         *issue*
>>>         A caller calls in
>>>
>>>         and then I place an outbound call and finally
>>>         bridge the two
>>>
>>>         calls.
>>>         This works as
>>>
>>>         expected, except when the outbound caller
>>>         hangs up first the
>>>
>>>         BYE never gets back to Asterisk.
>>>         I can see the BYE
>>>
>>>         reach OpenSips but a '404 not here' is
>>>         returned to the ISP.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>         sip trace https://gist.github.com/5009662
>>>
>>>
>>>
>>>
>>>
>>>
>>>         opensips.cfg https://gist.github.com/5009704
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>         thanks for your time.
>>>
>>>
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>>>         Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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>>
>>
>>
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>
>
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