[OpenSIPS-Users] 302 handling

Bogdan-Andrei Iancu bogdan at voice-system.ro
Tue Aug 19 12:30:04 CEST 2008


Hi Alex,

Glad you solved the problem - at least at signalling level :)

Do you have NAT involved ? Have you checked the SDP (Ip and port) in 
both request and reply to see if where the problem comes?

Regards,
Bogdan

Alex G wrote:
> well i did make some headway on this, unfortunately i had to get 
> tricky with it.
>
> Even with the get redirects, it was still not placing the correct 
> redirect in there. As a matter of fact,  it seems like the function 
> was not working at all in the failure_route. My solution involved 
> setting an avp in the reply route becuase both the source and 
> destination of the paceket were the same when it was in the failure 
> route.  So in on reply i set an avp that then if was true in the 
> branch route just rewrote the host port. So great I was able to make 
> the call path divert but when the 2 pstn endpoints actually link, 
> there is no sound. There seems to be rtp when i look in asterisk's 
> cli, but neither side is giving me audio  :(
>
> In the branch route, i tried with and without forecrtp proxy, but no 
> dice....
>
> anyone have an idea as to what might be going on?
>
> as always any input is greatly appreciated :)
>
>
> On Sun, Aug 17, 2008 at 2:16 PM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Hi Alex,
>
>     Actually, after the get_redirects(), you should not do a
>     rewiteXXXX() - just to t_relay(); the get_redirects() already
>     populates the new branch with all the information.
>
>     Regards,
>     Bogdan
>
>
>     Ovidiu Sas wrote:
>
>         If you want to rewrite the port, you need to use the following
>         syntax:
>         rewritehostport("XXX.XXX.XXX.XXX:ZZZZZ");
>         where ZZZZZ is the new port.
>
>
>         Regards,
>         Ovidiu Sas
>
>         On Wed, Aug 13, 2008 at 4:54 PM, Alex G
>         <greekman0000 at gmail.com <mailto:greekman0000 at gmail.com>> wrote:
>          
>
>             unfortunately the solution is a bit vague for what I'm
>             trying to do...
>
>
>             in the 302 packet the contact for redirect is sip
>             xyz at abc.abc.abc.abc
>
>             failure_route[1] {
>                if (t_check_status("302")) {
>                xlog("in redirect failure $fu");
>                 get_redirects("*:1","redirect");
>                  rewritehostport("XXX.XXX.XXX.XXX");
>                 t_relay();
>                }
>
>             this should take the contact address and rewrite the host
>             port for it
>             relaying it to the new location right? should be an
>             immediate invite to
>             abc at XXX.XXX.XXX.XXX
>
>             unfortunately it doesn't rewrite the host port. It merely
>             relays directly to
>             the contact in the 302 packet xyz at abc.abc.abc.abc
>
>             any ideas would be welcome :)
>
>             thanks
>
>             alex
>
>             On Wed, Aug 13, 2008 at 2:38 PM, Ovidiu Sas
>             <osas at voipembedded.com <mailto:osas at voipembedded.com>> wrote:
>                
>
>                 It is all here:
>                 http://www.opensips.org/html/uac_redirect.html#id2519995
>
>                 Regards,
>                 Ovidiu Sas
>
>                 On Wed, Aug 13, 2008 at 2:03 PM, Alex G
>                 <greekman0000 at gmail.com
>                 <mailto:greekman0000 at gmail.com>> wrote:
>                      
>
>                     I know there was some stuff about how to handle
>                     302s and send forward a
>                     new
>                     invite to the diversion contact on the old mailing
>                     list archives, but
>                     they
>                     are all gone now :(
>
>                     wondering if anyone can help me with this.....
>
>                     opensips -> ua -> moved -> opensips invite contact
>                     from diversion
>
>
>
>                     basically opensips makes an invite to locally
>                     registered uac, the uac
>                     redirects to an external pstn number XXX-XXX-XXXX,
>                     opensips then needs
>                     to
>                     handle the 302 and generate an invite to XXX-XXX-XXXX
>
>
>                     any help would be most appreciated
>
>                     thanks Alex
>
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>
>                            
>
>                
>
>
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>
>
>




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