#
# $Id: ser-isdngw.conf,v 1.2 2003/09/09 17:48:22 ullstar Exp $
#
# isdngw sample config script
# please direct comments to ullstar@iptel.org
#

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

# setup parameters according to your needs. Most people will only have 
# to adjust the listen and alias parameters below.

debug=1		        # debug level (cmd line: -dddddddddd)
fork=yes

log_stderror=no	# (cmd line: -E)
check_via=yes		# (cmd. line: -v)
dns=0			# (cmd. line: -r)
rev_dns=0		# (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"

# Add the name of your system here
#listen=yourhostname
# for more names add alias entries, all that might be used as domain in SIP URI
#alias=yourhostname.yourdomain.com
#alias=your.ip.addr.ess


# ------------------ module loading ----------------------------------

loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
loadmodule "/usr/local/lib/ser/modules/dbtext.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
	

# ----------------- setting module-specific parameters ---------------

# You may want to define things like databases here. Please refer to 
# the extensive SER documentation for this purpose. Module parameters 
# are always described in the modules README files.
#
# For pure isdn gateway functionality only a database is needed, we use
# a simple textfile for this purpose. Actually this is only neccessary until
# the vm module is reworked. Simply copy the etc/db directory from the isdngw
# directory somewhere and specify it in the following statement:

modparam("voicemail", "db_url","/root/ser/db")


# -------------------------  request routing logic -------------------

# This section describes how SIP messages are handled. 

# workaround needed for some buggy UAs (e.g. MS Messenger)
modparam("rr", "enable_full_lr", 1)

# The routing is described now:
route{

	# initial sanity checks -- messages with
	# max_forwars==0, or excessively long requests
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		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
	loose_route();

	# deal with my domain first
	if (uri==myself) {


		# the following allows for user-agent registration
		# in concerns of the isdn gateway this means:
		# register a SIP phone with <telephonenumber>@<this server name>
		# and configure the isdn gateway to listen on the number specified as 
		# the user (<telephonenumber>).
		# Incoming calls are then directed to this user agent
		# Note: This sample configuration does not use a persistent storage, so
		# if you restart SER you have to re-register your SIP client to make this
		# work. Refer to the userloc module's documentation for setting up 
		# persistent contact storage.
		if (method=="REGISTER") {
			save("location");
			break;
		};
		if (lookup("location")) {
			if (!t_relay()) {
				sl_reply_error();
			};
			break;
		};
		
		# ############################## #
		# isdngw specific configuration  #
		# ############################## #
		if(t_newtran()){

			if(method=="INVITE" || method=="BYE" || method=="CANCEL"){

				# send a response right at the start to avoid retransmissions
				t_reply("100","Trying -- just wait a minute !");

				# isdngw only gets activated on invite requests
				if(method=="INVITE"){
					
					# isdngw feels to be responsible for numeric userparts
					# all numbers followed by @ and anything after it match
					# this expression
					# for example: sip:555123123@yourdomain.com:5061 matches.
					# The vm command (from module vm) is used to contact the 
					# media server and though it the isdngw.
					# /tmp/am_fifo is the fifo filename ued for communications, make
					# shure the permissions are correct and that the same fifo
					# filename is defined in sems.conf.
					if(uri=~"sip:[0-9]+@.*"){
						if(!vm("/tmp/am_fifo","isdngw")){
							log("could not contact isdngw\n");
							t_reply("500","could not contact isdngw");
						};
					# we dont feel responsible for sip addresses not starting with
					# a number, so send the right error code.
					} else {
						t_reply("404","Not Found");
					};
					# stop routing here, the message is now processed by the media server
					break;
				};

				# The following handles the call termination, we must pass these requests
				# to the media server as follows. Again make shure the fifo name and permissions
				# are set correctly (like im sems.conf).
				if((method=="BYE")||(method=="CANCEL")){
						if(!vm("/tmp/am_fifo","bye")){
							log("could not contact the media server\n");
							t_reply("500","could not contact the media server");
						};
						break;
				};
			# other methods than INVITE, BYE and CANCEL are not handled by this SIP Server
			# so we sent an error message
			} else {
				log("ERROR: method not supported\n");
				t_reply("500", "sorry, method not supported");
			};	
		} else {
		# for any reason the transaction could not be created, send error code
		log("could not create new transaction\n");
		sl_send_reply("500","could not create new transaction");
		};
	# not uri=myself, this SIP request is not directed to us, simply direct it to its
	# correct destination
	} else {
		if (!t_relay()) {
			sl_reply_error();
		};
	};
# end of routing.
}