[Serusers] How to forward the call to asterisk voicemail box if no one answer the phone?

[gc]

I am using asterisk as my voicemail system and ser as my sip server. It works fine for a specific called number(4243) by using append_branch() function. But I don't know how to setup ser to forward any called number  to asterisk if no one answer the phone. I mean something like forward() function so I only need to specify the IP and port of asterisk.  

Here is my ser.cfg:


#
# $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=7

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)

modparam("tm", "fr_inv_timer", 15)
modparam("tm", "fr_timer", 10)

# -------------------------  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("seti", "subscriber")) {
    www_challenge("seti", "0");
    break;
   };

   save("location");
   break;
  };

  # native SIP destinations are handled using our USRLOC DB
  if (!lookup("location")) {
 #Handle PSTN calls.
   if (uri=~"^sip:8500 at .*")   #To asterisk voicemail admin.
   {
    record_route();
    rewritehostport("Asterisk server IP:PORT");
    forward(<Asterisk server IP:PORT>);
   }
   else
   {
    record_route();
    rewritehostport("PSTN IP:PORT");
    forward(PSTN IP:PORT);
   };
  };
 };
 t_on_failure("1");
 # 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();
 };

}
failure_route[1] {
 append_branch("sip:4243 at AsteriskIP:Port");
 t_relay();
}

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