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

Bogdan-Andrei IANCU iancu at fokus.fraunhofer.de
Tue Jun 15 10:51:00 CEST 2004


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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>  
>




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