[Serusers] outgoing calls with alias

[Ralph.Wabel at swisscom.com]

Hi,
 
I have the problem that when I make an outgoing call over a Cisco
Gateway to a PSTN phone. I've defined an alias for the user, works fine
for incoming calls, but for the outgoing calls it shows always the
number of the main number of the number block. When I make a debug I see
that it goes out with the username instead of the alias. Here is my
ser.cfg, maybe I've done something wrong in the config file. If someone
could help me it would be great. Let me know if the config from the
Cisco Gateway is also important.
 
# $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=no
#log_stderror=yes       # (cmd line: -E)
 
/* Uncomment these lines to enter debugging mode
debug=7
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/ser_fifo"
fifo_mode=0666
alias="swissptt.ch"
alias="inoitasip.swissptt.ch"
# ------------------ 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"
"/etc/ser/ser.cfg" 138L, 3639C
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");
                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;
        };
        # attempt handoff to PSTN
        if (uri=~"^sip:0[0-9]*@inoitasip.swissptt.ch") {  ##  This
assumes that the caller is
              log("Forwarding to PSTN\n");      ##  registered in our
realm
                t_relay_to_udp("193.5.228.202", "5060");  ##  Our Cisco
router
         break;
        forward( 193.5.228.202, 5060 );
         break;
        forward( 193.5.228.202, 5060 );
        };
 
        # 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=~inoitasip.swissptt.ch) {
 
                if (method=="REGISTER") {
 
        # Uncomment this if you want to use digest authentication
                        if (!www_authorize("inoitasip.swissptt.ch",
"subscriber")) {
                                www_challenge("inoitasip.swissptt.ch",
"0");
                                break;
                        };
 
                        save("location");
                        break;
                };
 
                #needed for alias
                lookup("aliases");
 
                # native SIP destinations are handled using our USRLOC
DB
                if (!lookup("location")) {
                        sl_send_reply("404", "Not Found");
                        break;
                };
        };
 
        # 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();
         break;
        forward( 193.5.228.202, 5060 );
        };
 
        # 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=~inoitasip.swissptt.ch) {
 
                if (method=="REGISTER") {
 
        # Uncomment this if you want to use digest authentication
                        if (!www_authorize("inoitasip.swissptt.ch",
"subscriber")) {
                                www_challenge("inoitasip.swissptt.ch",
"0");
                                break;
                        };
 
                        save("location");
                        break;
                };
 
                #needed for alias
                lookup("aliases");
 
                # native SIP destinations are handled using our USRLOC
DB
                if (!lookup("location")) {
                        sl_send_reply("404", "Not Found");
                        break;
                };
        };
 
        # 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();
        };
 
}
 
 
Thanks
 
Ralph
 
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