[Serusers] outgoing calls with alias

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