[OpenSIPS-Users] Check Live Peers on OpenSIPS

[Ahmed Munir]

Hi Bogdan,

Thanks for reply. I forgot to mention earlier that for I'm using OpenSIPS +
FreeRadius, where radius is doing accounting and authentication. I used
aaa_does_uri_exist() function as well, but seems not working or making
mistake while implementing it. On other hand using lookup("location",m)
function, on retcode = -1, I redirected the INVITE to GW, using Dispatcher.
 But though thanks for your suggestion and I'll consider it.

Few things I want to ask you, as I listed below;
1-How can I forward SIP INVITE request to other SIP machine in state full
manner ?
2- While accounting using radius, when user A (registered on OpenSIPS) calls
the user B who is located at GW side, accounting doesn't take place.  On the
other hand when user B (from GW) calls user A (to OpenSIPS), accounting take
place. I want to know its cause? Because I want its accounting on both
sides.

Kindly advise me at your earliest.


> ------------------------------
>
> Message: 6
> Date: Thu, 18 Mar 2010 10:23:27 +0200
> From: Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Message-ID: <4BA1E2FF.3060702 at voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed,
>
> if the destination number (called number) is not a local subscriber (a
> SIP user), you simply route the call to a PSTN GW (you do this re-route
> from the script)
>
> To check if a user is a local subscriber, you can either check a pattern
> (like all my local users are alphanumeric, or all starts with 3345*,
> etc), either simply check if the user does exists in the subscriber
> table (see the URI module, the db_does_uri_exists() function:
>    http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
> > Hi,
> >
> > I want to know how can I check the peers of source and destination
> > phones? Like if both phones are located (registered) on one
> > UAS(OpenSIPS) can call SIP-SIP, if any one phone is registered on UAS
> > and other is on PSTN, call will be re-routed to SIP-PSTN. In case of
> > SIP-SIP, lookup("location") function works and I need to know how can
> > I forward call to SIP-PSTN ?
> >
> > Kindly advise me the method/ function can used for it.
> >
> > --
> > Regards,
> >
> > Ahmed Munir
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
>
>
> --
Regards,

Ahmed Munir
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20100318/b0aa9f54/attachment.htm