Hi

 

here is my config

 

iam able to dial *86, i get voice message that no voice messages

 

But the call  rewriting when the user not available

it should go to asterisks voice mail

 

ram

 

[root@sert openser]# more openser.cfg
#
# $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $
#
# simple quick-start config script
#

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

debug=3            # debug level (cmd line: -dddddddddd)
log_facility=LOG_LOCAL7
fork=yes
log_stderror=no    # (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
fifo="/tmp/openser_fifo"

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

loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/acc.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/mysql.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/auth.so"
loadmodule "/usr/local/lib/openser/modules/auth_db.so"
loadmodule "/usr/local/lib/openser/modules/textops.so"
loadmodule "/usr/local/lib/openser/modules/uri.so"
loadmodule "/usr/local/lib/openser/modules/uri_db.so"
loadmodule "/usr/local/lib/openser/modules/group.so"
loadmodule "/usr/local/lib/openser/modules/msilo.so"
loadmodule "/usr/local/lib/openser/modules/nathelper.so"
loadmodule "/usr/local/lib/openser/modules/enum.so"
loadmodule "/usr/local/lib/openser/modules/domain.so"
loadmodule "/usr/local/lib/openser/modules/xlog.so"

 

fifo_db_url="mysql://openser:openserrw@localhost/openser"

modparam("usrloc|acc|auth_db|group|msilo", "db_url", "mysql://openser:openserrw@localhost/openser")

# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
/* 0 -- dont use mysql, 1 -- write_through, 2--write_back */
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 10)
modparam("usrloc", "use_domain", 1)
modparam("registrar", "use_domain", 1)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "use_rpid", 1)
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_prefix", "<sip:")
modparam("auth", "rpid_suffix", "@myip>;party=calling;id-type=subscriber;screen=yes;privacy=off")

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

# -- acc params --
# report ACKs too for sake of completeness -- as we account PSTN
# destinations which are RR, ACKs should show up
modparam("acc", "report_ack", 1)
modparam("acc", "log_level", 1)
# if BYE fails (telephone is dead, record-routing broken, etc.), generate
# a report nevertheless -- otherwise we would have no STOP event; => 1
#modparam("acc", "failed_transactions", 1)

# that is the flag for which we will account -- don't forget to
# set the same one :-)
# Usage of flags is as follows:
#   1 == should account(all to gateway),
#   3 == should report on missed calls (transactions to iptel.org's users),
#   4 == destination user wishes to use voicemail
#   6 == nathelper
#
modparam("acc", "log_flag", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "log_missed_flag", 3)
modparam("acc", "db_missed_flag", 3)

# report to syslog: From, i-uri, status, digest id, method
modparam("acc", "log_fmt", "fisumdpr")

# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 25)
modparam("tm", "wt_timer", 30)

# -- msilo params
modparam("msilo", "registrar", "sip:registrar@mydomain.com")

# -- enum params --
modparam("enum", "domain_suffix", "e164.arpa.")

# -- multi-domain
modparam("domain", "db_mode", 1)
# -------------------------  request routing logic -------------------

# main routing logic
#Routing Script
route {
 #check for old messages: could mean a problem with the DNS entries or some other loop-causer...
 if (!mf_process_maxfwd_header("10"))
 {
   xlog("L_WARN", "WARNING: Too many hops\n");
   sl_send_reply("483", "Too many hops, forward count exceeded limit\n");
   return;
 };

 #check for extremely large messages; we don't need a sip dos attack
 if (msg:len >= 2048)
 {
   xlog("L_WARN", "WARNING: Message too large, >= 2048 bytes\n");
   sl_send_reply("513", "Message too large, exceeded limit\n");
   return;
 };

 #record everything besides registers and acks
 if(method!="REGISTER" && method!="ACK")
 {
  setflag(1);
 };

 #do not send to voicemail if BYE or CANCEL
 #is used to end call before user pickup or timeout
 if(method=="CANCEL" || method=="BYE")
 {
  setflag(10);
 };

 #grant route if route headers already present
 if (loose_route())
 {
   route(1);
   return;
 };

 #Always require authentication, which could result in a PSTN, ie $$$
 if (method=="REGISTER")
 {
   if(!www_authorize("domain.com", "subscriber"))
   {
    www_challenge("domain.com", "0");
    return;
   }
   else
   {
    #Save into user database, used below when checking if user is available
    xlog("L_INFO", "REGISTER: User Authenticated Correctly\n");
    save("location");
    return;
   };
 };

 if (method=="INVITE")
 {
  if (uri=~"sip:1[0-9]+@.*")
  {
   #authorize if a call is going to PSTN
   if(!proxy_authorize("domain.com", "subscriber"))
   {
    proxy_challenge("domain.com", "0");
    return;
   };

   xlog("L_INFO", "CALL: Call to international number\n");
#   rewritehostport("voip_gw.domain.net:5060");
        rewritehostport("myprovider-voip:port");
  }
  else if(uri=~"sip:\*86@.*")
  {
   #authorize if a call is going to PSTN
   if(!proxy_authorize("domain.com", "subscriber"))
   {
    proxy_challenge(" domain.com", "0");
    return;
   };

   xlog("L_INFO", "CALL: Call to check voicemail\n");
   rewritehostport("asterisk-server-ip:5090");
  }
  else
  {
   if (does_uri_exist())

{
    #Call is to sip client, so do nothing but route
    xlog("L_INFO", "CALL: Sip client\n");
    if (!lookup("location"))
    {
     sl_send_reply("404", "Not Found");
     log(1, "ERROR: User Not Found\n");
     return;
    };
   }
   else
   {
    #authorize if a call is going to PSTN
    if(!proxy_authorize("domain.com ", "subscriber"))
    {
     proxy_challenge("domain", "0");
     return;
    };

    #Call destination is PSTN, so send it to the gateway (Net.com)
    xlog("L_INFO", "CALL: PSTN gateway\n");
#    rewritehostport("voip_gw.domain.net:5060");
        rewritehostport("myprovider-voip:port");

   };
  };

  #Make sure that all subsequent requests go through us;
  record_route();
 }
 else
 {
  if (does_uri_exist())
  {
   #Call is to sip client, so do nothing but route
   xlog("L_INFO", "CALL: Sip client\n");
   if (!lookup("location"))
   {
    sl_send_reply("404", "Not Found");
    log(1, "ERROR: User Not Found\n");

    return;
   };
  }
  else
  {
   #Call destination is PSTN, so send it to the gateway (Net.com)
   xlog("L_INFO", "CALL: PSTN gateway\n");
   rewritehostport("voip_gw.domain.net:5060");

  };
  record_route();
 };

 #ALL PROCESSING IS DONE, SO ROUTE
 route(1);
}

route[1]
{
#send the call outward

 if(method=="INVITE" && !isflagset(10))
 {
  t_on_failure("2");
 };

 if (!t_relay())
 {
  xlog("L_WARN", "ERROR: t_relay failed");
  sl_reply_error();
 };

}

failure_route[2]
{
 if(!t_was_cancelled())
 {
  revert_uri();
  rewritehostport("asterisk-ip-voicemail:5090");
  append_branch();
  #PREVENT SOME CRAZY VOICEMAIL LOOP
  xlog("L_INFO", "INFO: CALL TO VOICEMAIL");
  setflag(10);
  route(1);
 }
}

 

 

ram