Phone im trying to call is a regular phone hooked up to a PBX then PBX is
connected to a cisco router (acting as sip gateway), so the phone cannot
register with the server. How can I get around it as x-lite will only call
number(a)sipserv.foo.com. But only calls are being forwared to cisco gateway
is number(a)foo.com (dunno why). Please look at my ser.cfg and see what I can
change so calls to number(a)sipserv.foo.com will also be forwaded to cisco
gateway.
Thanks
Andy
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Wednesday, January 07, 2004 6:39 PM
To: Andy Singh
Cc: 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] dialing PSTN using x-lite
404 means that the phone you are trying to call is not registered on the
server. Make sure that you have proper domain in subscriber table
(
sipserv.foo.com and not just
foo.com).
Jan.
On 06-01 15:21, Andy Singh wrote:
Hello all,
My sip domain is
sipserv.foo.com, i have a user1(a)sipserv.foo.com. i can
log
in fine via messenger and via x-lite 2.0, but when i
dial a phone number
let's say 1212(a)sipserv.foo.com i immediatly get 404 not found. but i can
dial 1212(a)foo.com from windows messenger, since i don't have the option
to
specify just @foo.com in x-lite it always dials
1212(a)sipserv.foo.com. How can i make x-lite dial differently or how can i
make
sipserv.foo.com work. Here's my ser.cfg file
#
# $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
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=yes # (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)
# ------------------------- 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("sipserv.foo.com", "subscriber")) {
www_challenge("sipserv.foo.com", "0");
break;
};
save("location");
break;
};
# 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();
};
attempt handoff to PSTN
if (uri=~"^sip:3[0-9]*") { ## This assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in our realm
forward( 156.151.96.253, 5060 ); ## Our Cisco router
break;
};
}
Please help
Thanks
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