[OpenSIPS-Users] SIP Trunking

[Ghaith ALKAYYEM]

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
> >
> >
> >   
> 
> 
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