Hola, 
 
La razón por la que escribo esta vez es para ver si me pueden ayudar adirigir llamas hacia un voicemail configurado en Asterisk.
 
El escenario es el siguiente: hasta este punto tengo usuarios que se encuentran en LDAP, estos se autentican a través de un servidor Radius con OpenSER, una vez que tengo los usuarios en Openser estos pueden hacer llamadas SIP entre usuarios registrados asé como tambien realizar llamadas a la pstn utilizando para ello Asterisk como gateway, por otro lado los usuarios de OpenSER pueden consultar su buzón de voz al presionar *98, el buzón esta vacío porque no se como hacer que OpenSer redireccione la llamada a Asterisk en caso de que el cliente este ocupado o no disponible.
 
Se que tengo que usar el failure_route pero no se como aplicarlo,
 
Gracias en adelanto por la ayuda,
 
Mario F.
 
Mi openser.cfg es el siguiente:
 

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

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

check_via=no      # (cmd. line: -v)
dns=no          # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
port=5060
children=4
listen=udp:192.168.1.11
alias="tesis.com"
#
# uncomment the following lines for TLS support
#disable_tls = 0
#listen = tls:your_IP:5061
#tls_verify = 1
#tls_require_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem"
#tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem"
#tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem"

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

# Uncomment this if you want to use SQL database

loadmodule "/usr/local/lib/openser/modules/mysql.so"

loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/textops.so"
loadmodule "/usr/local/lib/openser/modules/avpops.so"
loadmodule "/usr/local/lib/openser/modules/xlog.so"
loadmodule "/usr/local/lib/openser/modules/uri.so"
loadmodule "/usr/local/lib/openser/modules/acc.so"
loadmodule "/usr/local/lib/openser/modules/auth_radius.so"
loadmodule "/usr/local/lib/openser/modules/group_radius.so"
loadmodule "/usr/local/lib/openser/modules/avp_radius.so"


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

###############PARAMETROS RADIUS
# -- acc params --

modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 2)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 1)
modparam("auth_radius", "service_type", 15)
modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp")
modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf")

# -- group_radius params --
modparam("group_radius", "use_domain", 1)

# -- avpops params --
modparam("avpops", "avp_aliases", "day=i:101;time=i:102")

# -- 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");
            exit;
      };

      if (msg:len >=  2048 ) {
            sl_send_reply("513", "Message too big");
            exit;
      };

      # 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())
      {
            append_hf("P-hint: rr-enforced\r\n");
            if(is_method("BYE"))
                  { # log it all the time
                          acc_rad_request("200 ok");
                          acc_log_request("200 ok");
                  }

            route(1);
      };


      if(is_method("INVITE") && !has_totag())
         
          {   # set the acc flags
           log(1,"-----> LLAMADA SIP <----- \n");
           setflag(1);
             setflag(2);           
          }
     
      if (uri==myself)
            {
                  if (method=="REGISTER")
                        {
                        if (!radius_www_authorize("tesis.com"))
                             
{
                              www_challenge("tesis.com", "0");
                              exit;
                                   
};
                        save("location");
                        exit;
                        };

         if(uri=~"sip:\*98@.*")
                {
                 #authorize if a call is going to PSTN
                xlog("L_INFO", "CALL: Call to check voicemail\n");
                rewritehostport("192.168.1.10:5060");
                };

                   if(uri=~"sip:041[2-6][0-9][0-9][0-9][0-9][0-9][0-9]+@")
                 
{
                  #xlog("L_ERR", "LLAMANDO A PSTN\n");      
                 
# set gateway address "ASTERISK"
                  log(1, "LLAMANDO A PSTN -----> Forwarding to Asterisk <----- \n");
                  rewritehostport("192.168.1.10:5060");
                        route(1);
                  };
       
                  lookup("aliases");
                  if (!uri==myself)
                         {
                        append_hf("P-hint: outbound alias\r\n");
                        route(1);
                         };

                  # native SIP destinations are handled using our USRLOC DB
                  if (!lookup("location"))
                        {
                        acc_rad_request("404 Not Found");
                              acc_log_request("404 Not Found");
                        sl_send_reply("404", "Not Found");
                        exit;
                        };
                  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();
      };
      exit;
}



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