[Serusers] PSTN Configuration
Abdul Lateef
abdul_zu at yahoo.com
Tue Jun 28 22:04:37 CEST 2005
Hello,
I read lot of example ducumentations from net about
the ser configuration. I found one example from
voip-info.org. i configured the ser.cfg file
accordidng to the guide. but when i am trying to
restart the ser using /usr/local/sbin/serctl restart i
am getting some pid file error messate like :
----------------------------------
[root at localhost root]# /usr/local/sbin/serctl stop
Stopping SER : No PID file found! SER probably not
running
[root at localhost root]# /usr/local/sbin/serctl start
Starting SER : PID file /var/run/ser.pid does not
exist -- SER start failed
[root at localhost root]#
--------------------------------
Here is my ser.cfg file configurations.
I wanted to explain how i am going to use it.
I have one VoIP Gatekeeper (MVTS) Which supports to
register SER i want to register the SER in MVTS and
the IP Phones i want to register in SER. So the call
can be pass to destination using like:
IP Phone > SER > MVTS > Destination PSTN.
MVTS IP : 195.22.146.20
SER : 10.0.0.20
IP Phone : Should use the ser IP and uid/pws
--------------- ser.cfg ----------------------
#
# $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $
#
#
# ------------------ module loading
----------------------------------
loadmodule "modules/sl/sl.so"
loadmodule "modules/tm/tm.so"
loadmodule "modules/acc/acc.so"
loadmodule "modules/rr/rr.so"
loadmodule "modules/maxfwd/maxfwd.so"
loadmodule "modules/mysql/mysql.so"
loadmodule "modules/auth/auth.so"
loadmodule "modules/auth_db/auth_db.so"
loadmodule "modules/group/group.so"
loadmodule "modules/uri/uri.so"
# ----------------- setting module-specific
parameters ---------------
modparam("auth_db",
"db_url","sql://ser:heslo@10.0.0.26/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# — acc params --
modparam("acc", "log_level", 1)
# that is the flag for which we will account — don't
forget to
# set the same one :-)
modparam("acc", "log_flag", 1 )
# ------------------------- request routing logic
-------------------
# main routing logic
route{
/* ********* ROUTINE CHECKS
********************************** */
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Wow — Message too
large");
break;
};
/* ********* RR
********************************** */
/* grant Route routing if route headers present
*/
if (loose_route()) { t_relay(); break; };
/* record-route INVITEs — all subsequent
requests must visit us */
if (method=="INVITE") {
record_route();
};
# now check if it really is a PSTN destination
which should be handled
# by our gateway; if not, and the request is an
invitation, drop it --
# we cannot terminate it in PSTN; relay
non-INVITE requests — it may
# be for example BYEs sent by gateway to call
originator
if (!uri=~"sip:\+?[0-9]+ at .*") {
if (method=="INVITE") {
sl_send_reply("403", "Call
cannot be served here");
} else {
forward(uri:host, uri:port);
};
break;
};
# account completed transactions via syslog
setflag(1);
# free call destinations ... no authentication
needed
if ( is_user_in("Request-URI", "free-pstn") /*
free destinations */
|
uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */
|
uri=~"sip:98[0-9][0-9][0-9][0-9]") {
log("free call");
} else if (src_ip==192.168.0.10) {
# our gateway doesn't support digest
authentication;
# verify that a request is coming from
it by source
# address
log("gateway-originated request");
} else {
# in all other cases, we need to check
the request against
# access control lists; first of all,
verify request
# originator's identity
if (!proxy_authorize( "gateway" /*
realm */,
"subscriber" /* table
name */)) {
proxy_challenge( "gateway" /*
realm */, "0" /* no qop */ );
break;
};
# authorize only for INVITEs —
RR/Contact may result in weird
# things showing up in d-uri that would
break our logic; our
# major concern is INVITE which causes
PSTN costs
if (method=="INVITE") {
# does the authenticated user
have a permission for local
# calls (destinations beginning
with a single zero)?
# (i.e., is he in the "local"
group?)
if (uri=~"sip:0[1-9][0-9]+ at .*")
{
if
(!is_user_in("credentials", "local")) {
sl_send_reply("403", "No permission for local calls");
break;
};
# the same for long-distance
(destinations begin with two zeros")
} else if
(uri=~"sip:00[1-9][0-9]+ at .*") {
if
(!is_user_in("credentials", "ld")) {
sl_send_reply("403", " no permission for LD ");
break;
};
# the same for international
calls (three zeros)
} else if
(uri=~"sip:000[1-9][0-9]+ at .*") {
if
(!is_user_in("credentials", "int")) {
sl_send_reply("403", "International permissions
needed");
break;
};
# everything else (e.g., interplanetary calls) is
denied
} else {
sl_send_reply("403",
"Forbidden");
break;
};
}; # INVITE to authorized PSTN
}; # authorized PSTN
# if you have passed through all the checks,
let your call go to GW!
rewritehostport("192.168.0.10:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
break;
};
}
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com
__________________________________
Yahoo! Mail Mobile
Take Yahoo! Mail with you! Check email on your mobile phone.
http://mobile.yahoo.com/learn/mail
More information about the sr-users
mailing list