[Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination

[Slot Zero]

Hello,

I am a Kamailio noob :). I am trying to get Asterisk to forward calls to 
my SIP provider via Kamailio. The same machine is running Kamailio and 
Asterisk. I do not want to consume credentials as they have to be passed 
on all the way to my SIP provider. There is no NAT of any sorts. SIP 
Phone/Users connect to Asterisk.I do not need to authenticate when forwarding call from Asterisk to Kamailio as they are 
both running on the same server but I do need to make sure that Kamailio 
dials and forwards 011+number to be sent from local host port 
5062(Asterisk listener) to SIP provider only. I have 6 Public IP addresses 
mapped on the server. I want to use the force_send_socket to allow me to 
change source IP of SIP requests when being sent to the SIP provider on the basis of credentials username in the request. I have pasted my config below. Please tell me what am I doing wrong here. In the kamctlrc file i have SIP_DOMAIN=localhost

Thank you

#------CONFIG BEGINS------------------
mpath="/lib/kamailio/modules_k/"

debug=3
fork=yes

children=4
auto_aliases=no
alias=localhost
alias=192.168.10.1
alias=192.168.10.2
alias=192.168.10.3
alias=192.168.10.4
alias=192.168.10.5
alias=192.168.10.6

disable_tcp=yes

loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "/lib/kamailio/modules/tm.so"
loadmodule "textops.so"

modparam("rr", "enable_full_lr", 1)

route {
        # Sanity Check
        # ------------

        # filter too old messages
        if (!mf_process_maxfwd_header("10")) {
                log("LOG: Too many hops\n");
                sl_send_reply("483","Too Many Hops");
                break;
        };


        if(msg:len>2048) {
               sl_send_reply("413", "message too large to be forwarded over UDP without fragmentation");
               exit;
        };

        # Record Route and NAT Preset
        # --------------------
        if (method != "REGISTER") {
                record_route();
        };

        # Loose Route

        # -----------
        if (loose_route()) {

                route(1);
                return;
        };

        # Call Type Processing
        # --------------------
        if (uri != myself) {
                route(1);
                return;
        };

        if (uri == myself) {
                if (method == "BYE") {
                        route(4);
                        return;
                } else if (method == "CANCEL") {
                        route(4);
                        return;
                } else if (method == "INVITE") {
                        route(3);
                        return;
                } else if (method == "NOTIFY") {
                        sl_send_reply("200", "Understood");
                        return;
                } else if (method == "OPTIONS") {
                        sl_send_reply("200", "Got it");
                        return;
                }
        };
        route(1);
}

# Default Message Handling
# -----------------------
route[1] {
       t_on_reply("1");
        if (!t_relay()) {
                sl_reply_error();
        };
}

# INVITE Message Handling
# ----------------------------------

# ----------------------------------
route[3] {
        if (uri =~ "^sip:011[0-9]@*") {
                rewritehostport("sip.voipprovider.com:5060");
                if (search("^(Contact|m): .*user01*@(127\.0\.0\.1|localhost)")) {
                force_send_socket(192.168.0.2:5060);
                };
                route(1);
                return;
        };

}

# CANCEL and BYE Message Handling
# ----------------------------------
route[4] {
        route(1);
}