Hi Sammy,
I was thinking to simplify a bit this and to add to LB module a function
like lb_is_from_peer() , to test if an IP + port matches on of the peers
defined in LB...
Regards,
Bogdan
On 04/04/2012 10:23 AM, SamyGo wrote:
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]
<mailto:[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).
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
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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