[OpenSIPS-Users] 302 handling
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Aug 20 14:14:02 CEST 2008
So, the first branch goes behind a NAT and gets redirected; the second
branch goes to PSTN and has the SDP from the first branch, right ?
get_redirects() works only in failure route.
Regards,
Bogdan
Alex G wrote:
> I did some more testing with this yesterday. The phone that is
> redirecting indeed is behind nat, but my pstn gateway is all open net.
> What I did find was that opensips was rediricting sdp to the phone
> still and not the gateway. Why there is still rtp traffic in asterisk
> is still a mystery, but I think with the right code I can also
> redirect the sdp out to the pstn as well.
>
> Will update you all on how this goes as I don't anticipate to start
> this scripting challenge till late in the week.
>
> A question that comes up is does the get_redirect function actually
> work in the failure route or am I misplacing it and it should be
> somewhere else?
>
> Thanks,
>
> Alex
>
> On Tue, Aug 19, 2008 at 6:30 AM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
> 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>
> <mailto: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>
> <mailto: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> <mailto: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>
> <mailto: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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