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

[gc]

I did read this part of example. It uses append_branch() function and you
have to specify the URI which include the dialed number to use it. Is there
any function like forward() I can use so I only need  to give the IP:PORT
and use whatever the dialed number in the original uri in falure_route[1]?

Gary
----- Original Message ----- 
From: "Bogdan-Andrei IANCU" <iancu at fokus.fraunhofer.de>
To: "gc" <garych at unidial.com>
Cc: <serusers at lists.iptel.org>
Sent: Tuesday, June 15, 2004 4:51 AM
Subject: Re: [Serusers] How to forward the call to asterisk voicemail box if
no one answer the phone?


> gc wrote:
>
> > 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.
>
> please see this first :
>
> http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSINGSECTION
>
> bogdan
>
> >
> > 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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> >serusers at lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
>