Hi Miha, I do exactly what Bogdan said, but using DB connecitons. IF its an Initial INVITE connect to DB and query table load_balancer and see if the source ip of INVITE matches any of the load_balanaced FS servers then I know that its from inside-media-serves to outside.
The next thing is identifying what original destination your FS was trying to send the calls to i.e carrier-ip / uri ! So in your FS dialplan add a sip header where you store the real destination of the SBC/trunk and once you are in the IF condition where you detect your internal FS servers, strip off that header change the $ru and t-relay the call !! phone(dial 007@FS )<===> FS (P-Real-DST: CAR.RIE.R.IP & dial 007@OpenSIPS)<====>OpenSIPS([email protected])<===> ITSP([email protected] ) I hope this is what you wanted. Regards, Sammy On Wed, Apr 4, 2012 at 11:49 AM, Miha <[email protected]> wrote: > Hi Bogan, > > that is a bit tricky as phones are registering on Opensips server. If I > make this that the phones will not register as FSs servers are on different > ips than SBC. > > What would you sugget? > > Regards, > Miha > > > On 4/2/2012 6:30 PM, Bogdan-Andrei Iancu wrote: > >> Hi Miha, >> >> Well, in your script, when dealing with the initial requests, just look >> at the source IP of the INVITEs - if from SBC, do the lb stuff, otherwise >> route it back to SBC. >> if (src_ip==11.22.33.44) { >> # do LB >> } else { >> # send to SBC >> } >> >> Regards, >> Bogdan >> >> On 04/02/2012 09:30 AM, Miha wrote: >> >>> Hi, >>> >>> as I am dealing with opensips for the first time I would ask you for a >>> little help. >>> I have installed and configured opensips that works like load_balancer ( >>> http://wiki.freeswitch.org/**wiki/Enterprise_deployment_**OpenSIPS<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> >>> ). >>> >>> I tested it and works. Than I have created siptrunk and point it to >>> Opensips. Opensips was balacing the calls to one of the FSs, that I have >>> set in opensips configuration. >>> >>> How can I now configure opensips, if the call is made from FS, that >>> opensips will send it to SBC (from where sip trunk is made), so that the >>> calls will be working in both direction? >>> >>> >>> Thanks! >>> >>> Miha >>> >>> ______________________________**_________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >>> >>> >> >> > > ______________________________**_________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >
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