etc/sip-router.cfg.m4
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 ### m4 macros to make the configuration easier
 
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 include(`rules.m4')
 
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 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')
 
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 declare(flags, ACC_FLAG, MISSED_FLAG, VM_FLAG, NAT_FLAG)
 declare(route, PSTN_ROUTE, NAT_ROUTE, VOICEMAIL_ROUTE, PSTN2_ROUTE)
 declare(onreply, NAT_REPLY)
 declare(failure, PSTN_FAILURE, _1_FAILURE)
 
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 ### End of m4 macro section
 
 #
 # $Id$
 #
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 # sip-router.cfg m4 template
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 #
 
 #
 # 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
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 fifo="/tmp/sip-router_fifo"
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 # uncomment to enter testing mode
 /*
 fork=no
 port=5064
 log_stderror=yes
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 fifo="/tmp/sip-router_fifox"
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  */
 
 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
 
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 loadmodule "/usr/local/lib/sip-router/modules/tm.so"
 loadmodule "/usr/local/lib/sip-router/modules/sl.so"
 loadmodule "/usr/local/lib/sip-router/modules/acc.so"
 loadmodule "/usr/local/lib/sip-router/modules/rr.so"
 loadmodule "/usr/local/lib/sip-router/modules/maxfwd.so"
 loadmodule "/usr/local/lib/sip-router/modules/mysql.so"
 loadmodule "/usr/local/lib/sip-router/modules/usrloc.so"
 loadmodule "/usr/local/lib/sip-router/modules/registrar.so"
 loadmodule "/usr/local/lib/sip-router/modules/auth.so"
 loadmodule "/usr/local/lib/sip-router/modules/auth_db.so"
 loadmodule "/usr/local/lib/sip-router/modules/textops.so"
 loadmodule "/usr/local/lib/sip-router/modules/uri.so"
 loadmodule "/usr/local/lib/sip-router/modules/group.so"
 loadmodule "/usr/local/lib/sip-router/modules/msilo.so"
 loadmodule "/usr/local/lib/sip-router/modules/nathelper.so"
 loadmodule "/usr/local/lib/sip-router/modules/enum.so"
 loadmodule "/usr/local/lib/sip-router/modules/domain.so"
 #loadmodule "/usr/local/lib/sip-router/modules/permissions.so"
 
 modparam("usrloc|acc|auth_db|group|msilo", "db_url", "sql://sip-router:heslo@localhost/sip-router")
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 # -- 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)
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 modparam("auth_db", "plain_password_column", "password")
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 #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") {
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                         route(PSTN_ROUTE); # Forward to PSTN gateway
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                 } else {
                         append_hf("P-hint: rr-enforced\r\n");
                         # account all BYEs 
                         if (method=="BYE") setflag(ACC_FLAG);
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                         route(NAT_ROUTE);  # Generic forward
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                 };
                 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");
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                 route(NAT_ROUTE);
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                 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");
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                 route(NAT_ROUTE);
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                 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]+@") {
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 			route(PSTN_ROUTE);
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 		} 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
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                 route(VOICEMAIL_ROUTE);
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                 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");
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                 route(PSTN_ROUTE);
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                 break;
         };
 
         # if user is on-line and is in voicemail group, enable redirection
         /* no voicemail currently activated
         if (method == "INVITE" && isflagset(VM_FLAG)) {
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                 t_on_failure(_1_FAILURE);    # failure_route() not defined
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         };
         */
 
         # ... 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");
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         route(NAT_ROUTE);
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 }
 
 #
 # Forcing media relay if necesarry
 #
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 route[NAT_ROUTE] {
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     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
 
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     t_on_reply(NAT_REPLY);
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     if (!t_relay()) {
             sl_reply_error();
             break;
     };
 }
 
 
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 onreply_route[NAT_REPLY] {
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         # 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();
         };
 
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         # keep Cisco gateway sending keep-alives
         if (isflagset(7) && status=~"2[0-9][0-9]") {   # flag(7) is mentioned NAT_FLAG ??
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                 remove_hf("Session-Expires");
                 append_hf("Session-Expires: 60;refresher=UAC\r\n");
                 fix_nated_sdp("1");
         };
 }
 
 
 #
 # logic for calls to the PSTN
 #
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 route[PSTN_ROUTE] {
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         # 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
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 			route(PSTN2_ROUTE);
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 			break;
 		};
 
 		if (uri =~ "^sip:00[1-9][0-9]+@") {
 			if (!is_user_in("credentials", "int")) {
 			    sl_send_reply("403", "International numbers not allowed");
 			    break;
 			};
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 			route(PSTN2_ROUTE);
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 		} else {
 			sl_send_reply("403", "Invalid Number");
 			break;
 		};
         }; # authorized PSTN
 	break;
 }
 
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 route[PSTN2_ROUTE] {
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 	rewritehostport("GW_IP_1:5060");
 	consume_credentials();
 	append_hf("P-Hint: GATEWAY\r\n");
 
 	# Try alternative gateway on failure
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 	t_on_failure(PSTN_FAILURE);
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         # Our PSTN gateway is symmetric and can handle direction=active flag
         # properly, therefore we don't have to use RTP proxy
 	t_relay();
 }
 
 
 
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 failure_route[PSTN_FAILURE] {
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 	rewritehostport("GW_IP_2:5060");
 	append_branch();
 	t_relay();	
 }
 
 
 # ------------- handling of unavailable user ------------------
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 route[VOICEMAIL_ROUTE] {
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         # 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");
         };
 }