[OpenSIPS-Users] 404 Not Here

[Bogdan-Andrei Iancu]

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.
>>
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20130226/ecbed310/attachment.htm>