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.
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@.*") #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@AsteriskIP:Port");
t_relay();
}