- ser.cfg template preprocessed by m4

etc/rules.m4

@@ -0,0 +1,29 @@
+#
+# $Id$
+#
+# (c) 2003 iptel.org
+#
+# Rules to process ser.cfg templates
+#
+
+define(`FROM_GW', `(_FROM_GW(1))')
+define(`_FROM_GW', `ifdef(`GW_IP_$1', `_FROM_GW(incr($1))(src_ip == GW_IP_$1)ifelse($1, 1, , ` || ')')')
+
+define(`TO_GW', `(@(_TO_GW(1))([;:].*)*)')
+define(`_TO_GW', `ifdef(`GW_IP_$1', `_TO_GW(incr($1))(patsubst(GW_IP_$1, `\.', `\\.'))ifelse($1, 1, , `|')')')
+
+define(`DIGEST_REALM', `SER_HOSTNAME')
+define(`SER_IP_REGEX', `patsubst(SER_IP, `\.', `\\.')')
+define(`SER_HOSTNAME_REGEX', `patsubst(SER_HOSTNAME, `\.', `\\.')')
+define(`SER_HOST_REGEX', `((SER_IP_REGEX)|(SER_HOSTNAME_REGEX))')
+
+define(`FROM_MYSELF', `(src_ip == SER_IP)')
+
+define(`ACC_FLAG', 1)
+define(`MISSED_FLAG', 3)
+define(`VM_FLAG', 4)
+define(`NAT_FLAG', 6)
+
+define(`PSTN', 3)
+define(`NAT', 1)
+define(`VOICEMAIL', 4)

etc/ser.cfg.m4

@@ -0,0 +1,532 @@
+### m4 macros to make the configuration easier
+
+define(`SER_IP', `192.168.0.1')
+define(`SER_HOSTNAME', `foo.bar')
+
+define(`GW_IP_1', `192.168.0.2')
+define(`GW_IP_2', `192.168.0.3')
+
+### End of m4 macro section
+
+#
+# $Id$
+#
+# ser.cfg m4 template
+#
+
+#
+# Set the following in your CISCO PSTN gateway:
+# sip-ua
+#   nat symmetric role passive
+#   nat symmetric check-media-src
+#
+fork=yes
+port=5060
+log_stderror=no
+fifo="/tmp/ser_fifo"
+
+# uncomment to enter testing mode
+/*
+fork=no
+port=5064
+log_stderror=yes
+fifo="/tmp/ser_fifox"
+ */
+
+debug=3
+memlog=4  # memlog set high (>debug) -- no final time-consuming memory reports on exit
+mhomed=yes
+listen=SER_IP
+alias="SER_HOSTNAME"
+check_via=yes
+dns=yes
+rev_dns=no
+children=16
+
+# if changing fifo mode to a more restrictive value, put
+# decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438
+fifo_mode=0666
+
+loadmodule "/usr/local/lib/ser/modules/tm.so"
+loadmodule "/usr/local/lib/ser/modules/sl.so"
+loadmodule "/usr/local/lib/ser/modules/acc.so"
+loadmodule "/usr/local/lib/ser/modules/rr.so"
+loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
+loadmodule "/usr/local/lib/ser/modules/mysql.so"
+loadmodule "/usr/local/lib/ser/modules/usrloc.so"
+loadmodule "/usr/local/lib/ser/modules/registrar.so"
+loadmodule "/usr/local/lib/ser/modules/auth.so"
+loadmodule "/usr/local/lib/ser/modules/auth_db.so"
+loadmodule "/usr/local/lib/ser/modules/textops.so"
+loadmodule "/usr/local/lib/ser/modules/uri.so"
+loadmodule "/usr/local/lib/ser/modules/group.so"
+loadmodule "/usr/local/lib/ser/modules/msilo.so"
+loadmodule "/usr/local/lib/ser/modules/nathelper.so"
+loadmodule "/usr/local/lib/ser/modules/enum.so"
+loadmodule "/usr/local/lib/ser/modules/domain.so"
+#loadmodule "/usr/local/lib/ser/modules/permissions.so"
+
+modparam("usrloc|acc|auth_db|group|msilo", "db_url", "sql://ser:heslo@localhost/ser")
+
+# -- usrloc params --
+/* 0 -- dont use mysql, 1 -- write_through, 2--write_back */
+modparam("usrloc", "db_mode", 2)
+modparam("usrloc", "timer_interval", 10)
+
+# -- auth params --
+modparam("auth_db", "calculate_ha1", yes)
+modparam("auth_db", "password_column", "password")
+#modparam("auth_db", "use_rpid", 1)
+modparam("auth", "nonce_expire", 300)
+modparam("auth", "rpid_prefix", "<sip:")
+modparam("auth", "rpid_suffix", "@GW_IP_3>;party=calling;id-type=subscriber;screen=yes;privacy=off")
+
+# -- rr params --
+# add value to ;lr param to make some broken UAs happy
+modparam("rr", "enable_full_lr", 1)
+
+# -- acc params --
+# report ACKs too for sake of completeness -- as we account PSTN
+# destinations which are RR, ACKs should show up
+modparam("acc", "report_ack", 1)
+modparam("acc", "log_level", 1)
+# if BYE fails (telephone is dead, record-routing broken, etc.), generate
+# a report nevertheless -- otherwise we would have no STOP event; => 1
+modparam("acc", "failed_transactions", 1)
+
+# that is the flag for which we will account -- don't forget to
+# set the same one :-)
+# Usage of flags is as follows:
+#   1 == should account(all to gateway),
+#   3 == should report on missed calls (transactions to iptel.org's users),
+#   4 == destination user wishes to use voicemail
+#   6 == nathelper
+#
+modparam("acc", "log_flag", ACC_FLAG)
+modparam("acc", "db_flag", ACC_FLAG)
+modparam("acc", "log_missed_flag", MISSED_FLAG)
+modparam("acc", "db_missed_flag", MISSED_FLAG)
+
+# report to syslog: From, i-uri, status, digest id, method
+modparam("acc", "log_fmt", "fisum")
+
+# -- tm params --
+modparam("tm", "fr_timer", 20)
+modparam("tm", "fr_inv_timer", 90)
+modparam("tm", "wt_timer", 20)
+
+# -- msilo params
+modparam("msilo", "registrar", "sip:registrar@SER_HOSTNAME")
+
+# -- enum params --
+modparam("enum", "domain_suffix", "e164.arpa.")
+
+# -- multi-domain
+modparam("domain", "db_mode", 1)
+
+# NAT features turned off -- smartnat available only in nat-capable release
+# We will you flag 6 to mark NATed contacts
+modparam("registrar", "nat_flag", NAT_FLAG)
+# Enable NAT pinging
+modparam("nathelper", "natping_interval", 15)
+# Ping only contacts that are known to be behind NAT
+modparam("nathelper", "ping_nated_only", 1)
+
+# ---------------------  request 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 (msg:len >= max_len) {
+                sl_send_reply("513", "Message too large");
+                break;
+        };
+
+        # special handling for natted clients; first, nat test is
+        # executed: it looks for via!=received and RFC1918 addresses
+        # in Contact (may fail if line-folding used); also,
+        # the received test should, if complete, should check all
+        # vias for presence of received
+        if (nat_uac_test("3")) {
+                # allow RR-ed requests, as these may indicate that
+                # a NAT-enabled proxy takes care of it; unless it is
+                # a REGISTER
+
+                if (method == "REGISTER" || !search("^Record-Route:")) {
+                        log("LOG: Someone trying to register from private IP, rewriting\n");
+
+                        # This will work only for user agents that support symmetric
+                        # communication. We tested quite many of them and majority is
+                        # smart smart enough to be symmetric. In some phones, like
+                        # it takes a configuration option. With Cisco 7960, it is
+                        # called NAT_Enable=Yes, with kphone it is called
+                        # "symmetric media" and "symmetric signaling". (The latter
+                        # not part of public released yet.)
+
+                        fix_nated_contact(); # Rewrite contact with source IP of signalling
+                        if (method == "INVITE") {
+                                fix_nated_sdp("1");  # Add direction=active to SDP
+                        };
+                        force_rport();       # Add rport parameter to topmost Via
+                        setflag(NAT_FLAG); # Mark as NATed
+
+                        append_to_reply("P-NATed-Caller: Yes\r\n");
+                };
+        };
+
+
+        # anti-spam -- if somene claims to belong to our domain in From,
+        # challenge him (skip REGISTERs -- we will chalenge them later)
+        if (search("(From|F):.*@SER_HOST_REGEX")) {
+                # invites forwarded to other domains, like FWD may cause subsequent 
+                # request to come from there but have iptel in From -> verify
+                # only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==fox)
+                if ((method == "INVITE" || method == "SUBSCRIBE") && !(FROM_MYSELF || FROM_GW)) {
+                        if  (!(proxy_authorize("DIGEST_REALM", "subscriber"))) {
+                                proxy_challenge("DIGEST_REALM", "0");
+                                break;
+                        };
+
+                        # to maintain outside credibility of our proxy, we enforce
+                        # username in From to equal digest username; user with
+                        # "john.doe" id could advertise "bill.gates" in From otherwise;
+                        if (!check_from()) {
+                                log("LOG: From Cheating attempt in INVITE\n");
+                                sl_send_reply("403", "That is ugly -- use From=id next time (OB)");
+                                break;
+                        };
+
+                        # we better don't consume credentials -- some requests may be
+                        # spiraled through our server (sfo@iptel->7141@iptel) and the
+                        # subsequent iteration may challenge too, for example because of
+                        # iptel claim in From; UACs then give up because they
+                        # already submitted credentials for the given realm
+                        #consume_credentials();
+                }; # non-REGISTER from other domain
+        } else if ((method == "INVITE" || method == "SUBSCRIBE" || method=="REGISTER" ) && 
+                   !(uri == myself || uri =~ "TO_GW")) {
+                # and we serve our gateway too (we RR requests to it, so that
+                # its address may show up in subsequent requests after loose_route
+                sl_send_reply("403", "No relaying");
+                break;
+        };
+
+        # By default we record route everything except REGISTERs
+        if (!(method=="REGISTER")) record_route();
+
+        # if route forces us to forward to some explicit destination, do so
+        #
+        # loose_route returns true in case that a request included
+        # route header fields instructing SER where to relay a request;
+        # if that is the case, stop script processing and just forward there;
+        # one could alternatively ignore the return value and treat the
+        # request as if it was an outbound one; that would not work however
+        # with broken UAs which strip RR parameters from Route. (What happens
+        # is that with two RR /tcp2udp, spirals, etc./ and stripped parameters,
+        # SER a) rewrites r-uri with RR1 b) matches uri==myself against RR1
+        # c) applies mistakenly user-lookup to RR1 in r-uri
+
+        if (loose_route()) {
+                # check if someone has not introduced a pre-loaded INVITE -- if so,
+                # verify caller's privileges before accepting rr-ing
+                if ((method=="INVITE" || method=="ACK" || method=="CANCEL") && uri =~ "TO_GW") {
+                        route(PSTN); # Forward to PSTN gateway
+                } else {
+                        append_hf("P-hint: rr-enforced\r\n");
+                        # account all BYEs 
+                        if (method=="BYE") setflag(ACC_FLAG);
+                        route(NAT);  # Generic forward
+                };
+                break;
+        };
+
+        # -------  check for requests targeted out of our domain... -------
+        if (!(uri == myself || uri =~ "TO_GW")) {
+                # ... and we serve our gateway too (we RR requests to it, so that
+                # its address may show up in subsequent requests after
+                # rewriteFromRoute
+                append_hf("P-hint: OUTBOUND\r\n");
+                route(NAT);
+                break;
+        };
+
+
+        # ------- now, the request is for sure for our domain -----------
+        # registers always MUST be authenticated to
+        # avoid stealing incoming calls
+        if (method == "REGISTER") {
+                /*
+                if (!allow_register("register.allow", "register.deny")) {
+                        log(1, "LOG: alert: Forbidden IP in Contact\n");
+                        sl_send_reply("403", "Forbidden");
+                        break;
+                };
+                */
+
+                # prohibit attempts to grab someone else's To address 
+                # using  valid credentials; 
+                if (!www_authorize("DIGEST_REALM", "subscriber")) {
+                        # challenge if none or invalid credentials
+                        www_challenge("DIGEST_REALM", "0");
+                        break;
+                };
+
+                if (!check_to()) {
+                        log("LOG: To Cheating attempt\n");
+                        sl_send_reply("403", "That is ugly -- use To=id in REGISTERs");
+                        break;
+                };
+
+                # it is an authenticated request, update Contact database now
+                if (!save("location")) {
+                        sl_reply_error();
+                };
+
+                m_dump();
+                break;
+        };
+
+        # some UACs might be fooled by Contacts our UACs generate to make MSN
+        # happy (web-im, e.g.) -- tell its urneachable
+        if (uri =~ "sip:daemon@") {
+                sl_send_reply("410", "Daemon is gone");
+                break;
+        };
+
+        # aliases
+        # note: through a temporary error in provisioning interface, there
+        # are now aliases 905xx ... they take precedence overy any PSTN numbers
+        # as they are resolved first
+        lookup("aliases");
+
+        # check again, if it is still for our domain after aliases
+        if (!(uri == myself || uri =~ "TO_GW")) {
+                append_hf("P-hint: ALIASED-OUTBOUND\r\n");
+                route(NAT);
+                break;
+        };
+
+	# Remove leading + if it is a number begining with +
+	if (uri =~ "^[a-zA-Z]+:\+[0-9]+@") {
+		strip(1);
+		prefix("00");
+	};		
+
+	if (!does_uri_exist()) {
+		# Try numeric destinations through the gateway
+		if (uri =~ "^[a-zA-Z]+:[0-9]+@") {
+			route(PSTN);
+		} else {
+			sl_send_reply("604", "Does Not Exist Anywhere");
+		};
+		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(VM_FLAG);
+        };
+
+        # native SIP destinations are handled using our USRLOC DB
+        if (!lookup("location")) {
+                # handle user which was not found
+                route(VOICEMAIL);
+                break;
+        };
+
+        # check whether some inventive user has uploaded  gateway
+        # contacts to UsrLoc to bypass our authorization logic
+        if (uri =~ "TO_GW") {
+                log(1, "LOG: Weird! Gateway address in UsrLoc!\n");
+                route(PSTN);
+                break;
+        };
+
+        # if user is on-line and is in voicemail group, enable redirection
+        /* no voicemail currently activated
+        if (method == "INVITE" && isflagset(VM_FLAG)) {
+                t_on_failure("1");
+        };
+        */
+
+        # ... and also report on missed calls ... note that reporting
+        # on missed calls is mutually exclusive with silent C timer
+        setflag(MISSED_FLAG);
+
+        # we now know we may, we know where, let it go out now!
+        append_hf("P-hint: USRLOC\r\n");
+        route(NAT);
+}
+
+#
+# Forcing media relay if necesarry
+#
+route[NAT] {
+    if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")) {
+            sl_send_reply("479", "We don't forward to private IP addresses");
+            break;
+    };
+    if (isflagset(NAT_FLAG)) {
+	    if (!is_present_hf("P-RTP-Proxy")) {
+            	force_rtp_proxy();
+		append_hf("P-RTP-Proxy: YES\r\n");
+	    };
+            append_hf("P-NATed-Calee: Yes\r\n");
+    };
+
+    # nat processing of replies; apply to all transactions (for example,
+    # re-INVITEs from public to private UA are hard to identify as
+    # natted at the moment of request processing); look at replies
+
+    t_on_reply("1");
+
+    if (!t_relay()) {
+            sl_reply_error();
+            break;
+    };
+}
+
+
+onreply_route[1] {
+        # natted transaction ?
+        if (isflagset(NAT_FLAG) && status =~ "(183)|2[0-9][0-9]") {
+                fix_nated_contact();
+                force_rtp_proxy();
+        # otherwise, is it a transaction behind a NAT and we did not
+        # know at time of request processing? (RFC1918 contacts)
+        } else if (nat_uac_test("1")) {
+                fix_nated_contact();
+        };
+
+        # keep Cisco gatweay sending keep-alives
+        if (isflagset(7) && status=~"2[0-9][0-9]") {
+                remove_hf("Session-Expires");
+                append_hf("Session-Expires: 60;refresher=UAC\r\n");
+                fix_nated_sdp("1");
+        };
+}
+
+
+#
+# logic for calls to the PSTN
+#
+route[PSTN] {
+
+        # discard non-PSTN methods
+        if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "OPTIONS" || method == "BYE")) {
+                sl_send_reply("500", "only VoIP methods accepted for GW");
+                break;
+        };
+
+        # turn accounting on
+        setflag(ACC_FLAG);
+
+        # continue with requests to PSTN gateway ...
+
+        # no authentication needed if the destination is on our free-pstn
+        # list or if the caller is the digest-less gateway
+        #
+        # apply ACLs only to INVITEs -- we don't need to protect other
+        # requests, as they don't imply charges; also it could cause troubles
+        # when a call comes in via PSTN and goes to a party that can't
+        # authenticate (voicemail, other domain) -- BYEs would fail then
+        if (method == "INVITE") {
+		if (!is_user_in("Request-URI", "free-pstn")) {
+                	if (!proxy_authorize("DIGEST_REALM", "subscriber"))  {
+                        	proxy_challenge("DIGEST_REALM", "0");
+                        	break;
+                	};
+
+                	# let's check from=id ... avoids accounting confusion
+                	if (!check_from()) {
+                        	log("LOG: From Cheating attempt\n");
+                        	sl_send_reply("403", "That is ugly -- use From=id next time (gw)");
+                        	break;
+                	};
+		} else {
+			# Allow free-pstn destinations without any checks
+			route(5);
+			break;
+		};
+
+		if (uri =~ "^sip:00[1-9][0-9]+@") {
+			if (!is_user_in("credentials", "int")) {
+			    sl_send_reply("403", "International numbers not allowed");
+			    break;
+			};
+			route(5);
+		} else {
+			sl_send_reply("403", "Invalid Number");
+			break;
+		};
+        }; # authorized PSTN
+	break;
+}
+
+route[5] {
+	rewritehostport("GW_IP_1:5060");
+	consume_credentials();
+	append_hf("P-Hint: GATEWAY\r\n");
+
+	# Try alternative gateway on failure
+	t_on_failure("7");
+        # Our PSTN gateway is symmetric and can handle direction=active flag
+        # properly, therefore we don't have to use RTP proxy
+	t_relay();
+}
+
+
+
+failure_route[7] {
+	rewritehostport("GW_IP_2:5060");
+	append_branch();
+	t_relay();	
+}
+
+
+# ------------- handling of unavailable user ------------------
+route[VOICEMAIL] {
+        # message store
+        if (method == "MESSAGE") {
+                if (!t_newtran()) {
+                        sl_reply_error();
+                        break;
+                };
+
+                if (m_store("0")) {
+                        t_reply("202", "Accepted for Later Delivery");
+                        break;
+                };
+
+                t_reply("503", "Service Unavailable");
+                break;
+        };
+
+        # non-Voip -- just send "off-line"
+        if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) {
+                sl_send_reply("404", "Not Found");
+                break;
+        };
+
+        if (t_newtran()) {
+                if (method == "ACK") {
+                        log(1, "CAUTION: strange thing: ACK passed t_newtran\n");
+                        break;
+                };
+
+                t_reply("404", "Not Found");
+        };
+
+        # we account missed incoming calls; previous statteful processing
+        # guarantees that retransmissions are not accounted
+        if (method == "INVITE") {
+                acc_log_request("404 missed call\n");
+                acc_db_request("404 missed call", "missed_calls");
+        };
+}