[OpenSIPS-Users] Limit calls over IP between Opensips Servers

[Bakko]

Hello,

maybe I resolved this configuration. My escenario:

- load balancing environment
- some clients register to opensips with a softphone and IP change.
- some clients sending only INVITE to Opensips and Asterisk Servers 
authenticate the INVITE.

Configuration:

I created a new table on Opensips database with this fields:
IP
Max calls permitted
company name

On both Opensips configuration I created a shared profile dialog:

Opensips 1
#### DIALOG module
loadmodule "dialog.so"
modparam("dialog","cachedb_url","redis://localhost:6379/")
modparam("dialog", "default_timeout", 21600)
modparam("dialog", "profiles_with_value", "calls/s")

Opensips2
loadmodule "dialog.so"
modparam("dialog","cachedb_url","redis://localhost:6380/")
modparam("dialog", "default_timeout", 21600)
modparam("dialog", "profiles_with_value", "calls/s")

On both record route:

record_route();

         setflag(1); # do accounting

                 if (load_balance("1","voip/s")) {
                 set_dlg_profile("calls/s","$si");
                 get_profile_size("calls/s","$si","$avp(size)");
                 avp_db_query("select canales from canales where 
IP='$si'","$avp(calls)");
                 if($retcode == 1){
                 xlog("L_INFO", "Canales activos = $avp(size) Canales 
disponibles = $avp(calls) $si\n");
                         if($avp(size) <= $avp(calls)){
                         xlog("L_INFO", "Llamada de $fu a $tu para $du \n");
                         route(RELAY);
                         exit;
                         }
                         else {
                         xlog("L_INFO", "Se ha superado el numero de 
canales disponibles [$avp(size)/$avp(calls)]\n");
                         route(2);
                         exit;
                         }
                 }
                 xlog("L_INFO", "Llamada de $fu a $tu para $du \n");
                 route(RELAY);
                 exit;
                 }


route[RELAY] {
         if (!t_relay()) {
                 xlog("L_ERR", "route [RELAY]\n");
                 sl_reply_error();

         };
         exit;
}

route[2] {
         xlog("desviando la llamada a contexto Opensip - extension 
canales\n");
         rewriteuri("sip:canales at sip.mydomain.net");
         forward();
         exit;
}

The logic:

when Opensips receive a INVITE set the profile size for the originating 
IP and get the profile size saving the value on the $avp(size) variable, 
then check the database to looking for for the IP on the INVITE. If IP 
exist, the function return 1 and save the max channels value on the 
$avp(calls) variable. If  $avp(size) is <= $avp(calls) opensips route 
the call to one of Asterisk servers else  process route(2). In process 
route[2] rewrite the URI and send the calls to one of Asterisk Servers 
where on context opensip, extension canales, a prompt announce to caller 
"no more channels availables" and hangup the call.

With this configuration I can share between the two opensips servers the 
MAX calls over IP permitted.

Maybe there is some syntax error on the script. Any suggestion?

Regards