Hi Ghaith, Force_send_socket() does not accept a variable,but you can use instead the $fs (force socket) var which does accept variables :
$var(a) = "x.x.x.x:xx"; $fs = $var(a) ; Regards, Bogdan Ghaith ALKAYYEM wrote: > 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 > > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users