Hello guys,
does anyone have a VEGA gateway, SER and Grandstream phones setup that
works? I am having some problems with outgoing calls. My call is being
dropped after 16 seconds, while the incoming call is working properly.
I have the latest firmware on my phones and here is my cnf file.
198.144.xxx.xxx - ser server
198.144.YYY.YYY - VEGA gateway
198.144.ZZZ.ZZZ - voicemail ( ser & sems )
#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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=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/textops.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"
loadmodule "/usr/lib/ser/modules/group.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("group", "db_url",
"mysql://ser:heslo@localhost/ser")
# time to give up on ringing -- global timer, applies to
# all transactions
modparam("tm", "fr_inv_timer", 20)
# ------------------------- request routing logic -------------------
# main routing logic
route {
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483", "Alas Too Many Hops");
break;
};
if (!(method=="REGISTER")) record_route();
if (loose_route()) {
t_relay();
break;
};
if (!uri==myself) {
t_relay();
break;
} else {
if (method == "REGISTER") {
#if (!save("location")) {
# sl_reply_error();
#};
# Uncomment this if you want to use digest
authentication
if (!www_authorize("198.144.xxx.xxx",
"subscriber")) {
www_challenge("198.144.xxx.xxx",
"0");
break;
};
save("location");
break;
};
};
# Destination PSTN or H323?
if( uri=~"^sip:9[0-9]*@198.144.xxx.xxx" )
{
route(1);
break;
};
if( uri=~"^sip:\*74@198.144.xxx.xxx" )
{
route(2);
break;
};
# does the user wish redirection on no availability? (i.e., is
he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
#if (is_user_in("Request-URI", "voicemail")) {
# setflag(4);
#};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
route(4);
break;
};
# if user is on-line and is in voicemail group, enable
redirection
if (method == "INVITE" ) {
t_on_failure("1");
};
t_relay();
}
# ------------ Send it to our PSTN ----------------------
route[1] {
# Route to PSTN Gateways(s)
if (uri=~"^sip:9[0-9]*@198.144.xxx.xxx") { ## This
assumes that th
e caller is
log("Forwarding to PSTN\n"); ## registered
in our re
alm
strip(1);
t_relay_to_udp( "198.144.YYY.YYY", "5060" );
break;
};
}
route[2] {
if (uri=~"^sip:\*74@198.144.xxx.xxx") { ## This assumes
that the c
aller is
log("Picking up a Call on PSTN\n"); ##
registered in
our realm
t_relay_to_udp( "198.144.YYY.YYY", "5060" );
break;
};
}
# ------------- handling of unavailable user ------------------
route[4] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# not voicemail subscriber
#if (!isflagset(4)) {
# sl_send_reply("404", "Not Found and no voicemail turned
on");
# break;
#};
# forward to voicemail now
rewritehostport("198.144.ZZZ.ZZZ:5090");
t_relay_to_udp("198.144. ZZZ.ZZZ", "5090");
#t_relay_to_tcp ("198.144. ZZZ.ZZZ","5090");
}
# if forwarding downstream did not succeed, try voicemail running
# at 198.144. ZZZ.ZZZ:5090
failure_route[1] {
revert_uri();
rewritehostport("198.144. ZZZ.ZZZ:5090");
append_branch();
t_relay_to_udp("198.144. ZZZ.ZZZ", "5090");
#t_relay_to_tcp ("198.144. ZZZ.ZZZ","5090");
}
Srbo Cvetkovic | CityNet, Inc.
srbo(a)city-net.com | Pittsburgh, PA
voice: 412.481.5406 | fax: 412.431.1315