#
# $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $
#
# config script with voicemail, PSTN dial-out functionality
#

# ----------- global configuration parameters ------------------------

debug=1         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no	# (cmd line: -E)

# Uncomment these lines to enter debugging mode 
/*
debug=8
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"
sip_warning=no
#
# ------------------ module loading ----------------------------------
#
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
#
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
#
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
#loadmodule "/usr/local/lib/ser/modules/uri.so"
#loadmodule "/usr/local/lib/ser/modules/group.so"
#
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/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")
#
#
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 2)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_fmt", "fimos")
#
modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 2)
modparam("acc", "db_missed_flag", 2)
#
#modparam("tm", "fr_inv_timer", 50)  #INVITE timeout
#modparam("tm", "fr_timer", 35)  #negative INVITE reply or no 
				#final reply for a request for ACK
#
modparam("voicemail", "db_url", "sql://ser:heslo@localhost/ser")
#
# -------------------------  request routing logic -------------------
#
# main routing logic
#
alias=10.10.10.49		#sip server IP address
alias=serserver			#sip server name
alias=mydomain.com		#sip domain/realm	
alias=serserver.mydomain.com	#sip server FQDN
#
route{
#	log(1,"entering main route");
	setflag(2); 	#set flag for accounting

	# 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 (len_gt( 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") {
			# digest authentication
#			log(1,"request for registration");
#			if (!www_authorize("mydomain.com", "subscriber")) {
#				www_challenge("mydomain.com", "0");
#				break;
#			};
			save("location");
			break;
		};

/* ********** Dial out to PSTN logic ************* */		

		#forward 411[information] and 911[emergency] requests to gateway
		if(uri=~"^sip:(4|9)11@(mydomain\.com|10\.10\.10\.49)"){
			log(1,"411/911 expression match");
			route(2);
			break;
		};
		#forward numerical 7 digit requests to gateway
		if(uri=~"^sip:[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
			log(1,"7 digit expression match");	
			route(2);
			break;    
		};
		# strip 650 and forward to GW if user dials 650 before phone no.
		if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
			strip(3);
			log(1,"650 area code dialed, 650 stripped");
			route(2);
			break;
		};
		#forward numerical 10 digit requests to gateway, append a 1 first
		if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
			prefix("1");
			log(1,"10 digit expression match, prefix 1");
			route(2);
			break;
		};
		#forward numerical 11 digit requests that start with a 1 to GW
		if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
			log(1,"10 digit exp match w/leading 1");
			route(2);
			break;
		};
		#forward international N digit requests to gateway
		if(uri=~"^sip:011[0-9]+@(mydomain\.com|10\.10\.10\.49)"){
			log(1,"international expression match");
			route(2);
			break;
		};

/* ********** VOICEMAIL logic ************* */
		if (uri=~"^sip:voicemail\+@"){
			log(1,"sip:voicemail uri match");
			route(3);
			break;
		};

/*  ****** Find Aliases and Locations of users ********* */
		#lookup "aliases" before looking up "location"
		lookup("aliases");

                # native SIP destinations are handled using our USRLOC DB
                if (!lookup("location")) {
			route(3);
			break;
		# this section needs help. I need to differentiate between users that
		# exits buy aren't online (send these requests to voicemail)
		# requests for users that aren't in my subscriber database
		# (send these requests a 404 reply)
                       	sl_send_reply("404", "User Not Found");
			break;
                };
	};

	# forward to current uri now; use stateful forwarding; that
	# works reliably even if we forward from TCP to UDP

	t_on_failure("1");

	if (!t_relay()) {
		sl_reply_error();
	};
	log(1,"route[]:eof");
}

route[2]{	
	log(1,"route[2]:SIP-to-PSTN call routed");
	rewritehostport("10.10.10.5:5060");
	if(!t_relay()){
		sl_reply_error();
	};
}

route[3]{
	log(1,"route[3]:vm:1");
	if (method=="INVITE" || method=="BYE" || method=="REFER"){
		log(1,"route[3]:vm:2");

		if(t_newtran()){
			t_reply("100","Trying -- just a second");
			if(method=="INVITE" || method=="REFER"){
				log(1,"route[3]:method==INVITE || REFER");
				if(uri =~ "conference" ){
					if(!vm("/tmp/am_fifo","conference")){
						log(1,"route[3]:vm:conference failed");
						t_reply("500","could not contact conference server");
					};
				}
				else if (uri =~"echo"){
					if(!vm("/tmp/am_fifo","echo")){
						log(1,"route[3]:vm:echo failed");
						t_reply("500","could not contact echo");
					};
				}
				else{
					if(!vm("/tmp/am_fifo","voicemail")){
						log(1,"route[3]:vm:voicemail failed");
						t_reply("500", "voicemail error");	
					};
				};
				break;
			};
			if(method=="BYE"){
				log(1,"route[3]:vm:method==BYE");
				if(!vm("/tmp/am_fifo","bye")){
					log(1,"route[3]:vm:bye failed");
					t_reply("500" , "could not contact the media server");
				};
				break;
			};
		}
		else{
			log(1,"route[3]:vm:new transaction failed");
			sl_send_reply("500", "new transaction failed");
			break;
		};
	};
}

route[4]{
	# this should be voicemail logic that is specific to a failure_route
	# i.e. line busy, or timeout after a certain period with no answer
	if(method=="INVITE" || method=="REFER"){
		if(!vm("/tmp/am_fifo","voicemail")){
			log(1,"route[3]:vm:voicemail failed");
			t_reply("500", "voicemail error");	
		};
	}else if(method=="BYE"){
		log(1,"route[3]:vm:method==BYE");
		if(!vm("/tmp/am_fifo","bye")){
			log(1,"route[3]:vm:bye failed");
			t_reply("500" , "could not contact the media server");
		};
	};

}

failure_route[1]{
	
	log(1,"failure_route[1]:jump to route[3]:vm");
#	append_branch("sip:info@mydomain.com");
	route(4);
}