[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
>
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> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
>
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