[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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