Hi I am having trouble with calls.

 

I have three clients created with alias’s       asterisk = 38212352, user1 = 38212351 , user2 =38212350

All users can register and I can see with serctl ul show the aliais and the user names.

I am using MYSQL and SERWEB also.

When I try to call the alias from each user 1 & 2 nothing happens. I think I have to correct something in the route section.

I want users to be able to call by alias and calls for PSTN to route to Asterisk with 0 prefix.

I start ser with /usr/local/sbin/ser -D –E

 

Can someone please help put my brain to rest.

Thank you for all your support.

 

 

 

 

# -------------------------  request routing logic -------------------

 

# main routing logic

 

route{

 

      # initial sanity checks -- messages with

      # max_forwards==0, or excessively long requests

      if (!mf_process_maxfwd_header("10")) {

            sl_send_reply("483","Too Many Hops");

            break;

      };

      if (msg:len >=  max_len ) {

            sl_send_reply("513", "Message too big");

            break;

      };

     

      # we record-route all messages -- to make sure that

      # subsequent messages will go through our proxy; that's

      # particularly good if upstream and downstream entities

      # use different transport protocol

      if (!method=="REGISTER") record_route();    

 

      # subsequent messages withing a dialog should take the

      # path determined by record-routing

      if (loose_route()) {

            # mark routing logic in request

            append_hf("P-hint: rr-enforced\r\n");

            route(1);

            break;

      };

 

      if (!uri==myself) {

            # mark routing logic in request

            append_hf("P-hint: outbound\r\n");

            route(1);

            break;

      };

 

      # if the request is for other domain use UsrLoc

      # (in case, it does not work, use the following command

      # with proper names and addresses in it)

      if (uri==myself) {

 

            if (method=="REGISTER") {

 

# Uncomment this if you want to use digest authentication

                  if (!www_authorize("192.168.1.4", "subscriber")) {

                        www_challenge("192.168.1.4", "0");

                        break;

                  };

 

                  save("location");

                  break;

            };

 

            lookup("aliases");

            if (!uri==myself) {

                  append_hf("P-hint: outbound alias\r\n");

                  route(1);

                  break;

            };

 

            # native SIP destinations are handled using our USRLOC DB

            if (!lookup("location")) {

                  sl_send_reply("404", "Not Found");

                  break;

            };

      };

      append_hf("P-hint: usrloc applied\r\n");

      route(1);

}

 

route[1]

{

      # send it out now; use stateful forwarding as it works reliably

      # even for UDP2TCP

      if (!t_relay()) {

            sl_reply_error();

      };

 

if (method == "INVITE" && (uri=~"^sip:0")){
rewritehostport ("
192.168.1.5:5060");
t_relay();
break();


}