Hi,
Is it possible also to make bridging dependent on a variable value by
passing a variable as a parameter to force_send_socket() as following:
$var(a) = "x.x.x.x:xx";
force_send_socket("$var(a)");
because the above configuration gave me an error but when I used the
variable in xlog function it was okay:
xlog("$var(a)");
I might do some code modification in this regard.
Regards.
On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote:
> Hi Matthew,
>
> There 2 things when comes bridging:
>
> 1) signalling part - selecting the proper outbound interface (private or
> public)
> a) this can be automatically done by opensips (based on the
> destination IP) if you enable the mhomed parameter in core ; this is
> simple by not so efficient
>
> b) you can do it manually, by selecting from script the correct
> interface - see the force_send_socket() function
>
> 2) media part
> a) rtpproxy - when enabling RTPproxy (at request and reply time)
> you can explicitly select which interface to use (see the e and i flags
> - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)
>
>
> Best regards,
> Bogdan
>
> Matthew S. Crocker wrote:
> > Hello,
> >
> > I'm brand new to OpenSIPS, just going through the make process now.
> >
> > I need to configure OpenSIPS to act like a SBC for some SIP trunks coming
> > off a VoIP switch. Where should I look for Documentation/Examples of a
> > working config?
> >
> > Here is my scenario:
> >
> > OpenSIPS has two interfaces, private & public.
> > VoIP Gateway is on private LAN with no gateway configured (it can only talk
> > to local machines, no routing)
> >
> > End user has an Asterisk server on a private lan behind their firewall (NAT)
> >
> > I need to configure OpenSIPS to listen for SIP messages on :5060 from the
> > end user firewall. It then need to rewrite the SIP message and send it to
> > the Gateway. The Gateway would see the messages coming from the internal
> > IP of the OpenSIPS server. Once all of the SIP messages get processed I
> > then need the OpenSIPS server to proxy the RTP streams (plan on using
> > mediaproxy) between the Asterisk server and VoIP Gateway.
> >
> > Any helpful hints on where to look?
> >
> > -Matt
> >
> >
> >
>
>
> _______________________________________________
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
