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@localhost root]# /usr/local/sbin/serctl stop Stopping SER : No PID file found! SER probably not running [root@localhost root]# /usr/local/sbin/serctl start
Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed [root@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]+@.*") { 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]+@.*") { 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]+@.*") { 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]+@.*") { 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