[SR-Users] Load balancing traffic based on SIP URI

Alberto Sagredo alberto.sagredo at avanzada7.com
Fri Apr 1 13:42:41 CEST 2016


I have done something similar as follows


if($rU=~"^[0-3][0-9][0-9]$")
{
$var(valor)=1;
}

And later..

if(!ds_select_domain("$var(valor)", "4")) {
                sl_send_reply("500", "Service Unavailable");
                xlog("L_INFO","[$fU@$si:$sp]{$rm} Sin destinos disponibles
para $rd \n");

                exit;
        }

That could be one idea to do that...

On your load balance table you have to use 0 -> All

1 -> Destination 1

2 -> Destination 2

an so.




2016-04-01 12:26 GMT+02:00 NITESH BANSAL <nitesh.bansal at outlook.com>:

> Hello,
>
> I want to use Kamailio to load balance traffic across multiple asterisks.
> The problem is that my SIP traffic is related to conference, so if a SIP
> call
> for a particular URI is sent to an Asterisk instance, any subsequent calls
> for that
> SIP URI should be sent to the same Asterisk instance.
> Is there anyway to achieve this in Kamailio?
>
> Thanks,
> Nitesh
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160401/25c0a893/attachment.html>


More information about the sr-users mailing list