[Serusers] Using AS5300 as PSTN Gateway

Ezequiel Colombo ecolombo at arcotel.net
Fri May 28 22:28:43 CEST 2004


Please make sure that the call is forwarded to your 5300. The if sentence must be (uri=~"^sip:1.*").
Also you can make a "debug ccsip message" in the gateway to see if the SIP INVITE comes in.
  ----- Original Message ----- 
  From: gc 
  To: serusers at lists.iptel.org 
  Sent: Friday, May 28, 2004 4:51 PM
  Subject: [Serusers] Using AS5300 as PSTN Gateway


  I have a AS5300 setup as PSTN Gateway. It works fine with VOCAL.
  Now I'd like to connect SER to this PSTN gateway.
  I added following line to the default ser.cfg file:
  if (uri=~"^sip:1") {
      log(1, "Forwarding to PSTN\n"j);
      forward(189.101.110.132, 5060);
      break;
  };

  This will allow any dialed number starting with 1 being forward to PSTN gateway.
  But it always give me busy signal.
  Can anybody tell me what's wrong?


  Here is my whole ser.cfg file:

  #
  # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
  #
  # simple quick-start config script
  #

  # ----------- global configuration parameters ------------------------

  #debug=3         # debug level (cmd line: -dddddddddd)
  #fork=yes
  #log_stderror=no # (cmd line: -E)

  /* Uncomment these lines to enter debugging mode 
  fork=no
  log_stderror=yes 
  debug=3
  */

  check_via=no # (cmd. line: -v)
  dns=no           # (cmd. line: -r)
  rev_dns=no      # (cmd. line: -R)
  #port=5060
  #children=4
  fifo="/tmp/ser_fifo"

  # ------------------ module loading ----------------------------------

  # Uncomment this if you want to use SQL database
  loadmodule "/usr/lib/ser/modules/mysql.so"

  loadmodule "/usr/lib/ser/modules/sl.so"
  loadmodule "/usr/lib/ser/modules/tm.so"
  loadmodule "/usr/lib/ser/modules/rr.so"
  loadmodule "/usr/lib/ser/modules/maxfwd.so"
  loadmodule "/usr/lib/ser/modules/usrloc.so"
  loadmodule "/usr/lib/ser/modules/registrar.so"

  # Uncomment this if you want digest authentication
  # mysql.so must be loaded !
  loadmodule "/usr/lib/ser/modules/auth.so"
  loadmodule "/usr/lib/ser/modules/auth_db.so"

  # ----------------- setting module-specific parameters ---------------

  # -- usrloc params --

  modparam("usrloc", "db_mode",   0)

  # Uncomment this if you want to use SQL database 
  # for persistent storage and comment the previous line
  modparam("usrloc", "db_mode", 2)

  # -- auth params --
  # Uncomment if you are using auth module
  #
  modparam("auth_db", "calculate_ha1", yes)
  #
  # If you set "calculate_ha1" parameter to yes (which true in this config), 
  # uncomment also the following parameter)
  #
  modparam("auth_db", "password_column", "password")

  # -- rr params --
  # add value to ;lr param to make some broken UAs happy
  modparam("rr", "enable_full_lr", 1)

  # -------------------------  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
   record_route(); 
   # loose-route processing
   if (loose_route()) {
    t_relay();
    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("vocal0", "subscriber")) {
      www_challenge("vocal0", "0");
      break;
     };

     save("location");
     break;
    };

    # native SIP destinations are handled using our USRLOC DB
    if (!lookup("location")) {
     sl_send_reply("404", "Not Found");
     break;
    };
   };
  #Handle PSTN calls.
   if (uri=~"^sip:1") {
    log(1,"Forwarding to PSTN\n");
    forward(189.101.110.132, 5060);
    break;
   };

   # forward to current uri now; use stateful forwarding; that
   # works reliably even if we forward from TCP to UDP
   if (!t_relay()) {
    sl_reply_error();
   };

  }




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