Re: [Serusers] Help when make a Call Busy Forward Failed? Does 'bestpractice' release??

24 Feb 2005


      Dear Greger:
I call t_on_failure("1"); on my route[2] (this is SIP to SIP route).
If I make a SIP to SIP call failed, it should go to another route.
For example:
  UA 1011 call 1022, and 1022 setting busy forward to PSTN(0939749xxx).
So, first route is route[2] (I setting t_on_failure("1") here), then
if busy condiction,
it will go to route[3](for PSTN).
I still can not understand where to add what you post before. So I add
my complete ser.cfg in this post.
Thank you for your kind.
Best Regard
Charles
ser.cfg:
---------------------------------------------------------------------------------------------------------
#
# $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=10         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no	# (cmd line: -E)
/* Uncomment these lines to enter debugging mode 
fork=no
log_stderror=yes
*/
listen=xxx.xxx.190.248
alias=ser.xxx.net.tw
alias=ser
alias=xxx.xxx.190.248
check_via=no		# (cmd. line: -v)
dns=no					# (cmd. line: -r)
rev_dns=no			# (cmd. line: -R)
port=5060
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# mhomed -- enable calculation of outbound interface.
# useful on multihomed servers.
mhomed=0
# ------------------ 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/textops.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/speeddial.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/pdt.so"
loadmodule "/usr/local/lib/ser/modules/options.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
# ----------------- setting module-specific parameters ---------------
# -- mediaproxy params --
modparam("mediaproxy", "natping_interval", 30)
modparam("mediaproxy", "sip_asymmetrics",
"/usr/local/etc/ser/sip-asymmetric-clients")
modparam("mediaproxy", "rtp_asymmetrics",
"/usr/local/etc/ser/rtp-asymmetric-clients")
# -- usrloc params --
#
# 0 = disable
# 1 = write-through
# 2 = write-back
#
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "desc_time_order", 1)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- db_url params --
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|pdt",
"db_url", "mysql://ser:heslo@localhost/ser")
# -- use_domain params --
modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0)
# -- permissions params --
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
# -- accounting params --
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "log_level", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 0)
# -- domain params --
modparam("domain", "db_mode", 1)
# ------------- exec parameters
modparam("exec", "setvars", 1)
modparam("exec", "time_to_kill", 10)
# -- registration params --
modparam("registrar", "nat_flag", 2)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 86400)
modparam("registrar", "default_expires", 3600)
modparam("registrar", "append_branches", 1)
# -- avp params --
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
#modparam("avpops", "use_domain", "1")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# To use more than one tables example
#modparam("avpops", "db_scheme",
"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name")
# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 22)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
# -- pdt params --
modparam("pdt", "db_table", "prefix_domain")
modparam("pdt", "prefix", "")
modparam("pdt", "hsize_2pow", 2)
modparam("pdt", "sync_time", 300)
modparam("pdt", "clean_time", 600)
# -- logging params
modparam("xlog", "buf_size", 8192)
# -- group params --
modparam("group", "table", "grp") 
modparam("group", "user_column", "username") 
modparam("group", "domain_column", "domain") 
modparam("group", "group_column", "grp")
# -- speeddial params --
modparam("speeddial", "user_column", "username")
modparam("speeddial", "domain_column", "domain")
modparam("speeddial", "sd_user_column", "username_from_req_uri")
modparam("speeddial", "sd_domain_column", "domain_from_req_uri")
modparam("speeddial", "new_uri_column", "new_request_uri")
modparam("speeddial", "domain_prefix", "tel")
#modparam("auth", "rpid_prefix", "Antek SIP <sip:")
#modparam("auth", "rpid_suffix",
"@ser.xxx.net.tw;user=phone>;party=calling;screen=no;privacy=on")
# -------------------------  request routing logic -------------------
# main routing logic
route {
# ------------------------------------------------------------------------
    # Sanity Check Section
    # ------------------------------------------------------------------------
    if (!mf_process_maxfwd_header("10")) {
    	sl_send_reply("483", "Too Many Hops");
    	break;
    };
if (msg:len > max_len) {
    	sl_send_reply("513", "Message Overflow");
    	break;
    };
# ------------------------------------------------------------------------
    # NOTIFY Keep-Alive Section
    # ------------------------------------------------------------------------
    if ((method=="NOTIFY") && search("^Event: keep-alive")) {
    	sl_send_reply("200", "OK");
    	break;
    };
#if ((method=="NOTIFY") && (uri=~"^sip:700@")) {
    #	sl_send_reply("200", "OK");
    #	break;
    #};
# ------------------------------------------------------------------------
    # NOTIFY Message-Summary Section
    #
    # Event: Message-Summary is used to control the message waiting indicator
    # ------------------------------------------------------------------------
    #if ((method=="NOTIFY") && (src_ip=="99.99.99.100") &&
search("^Event: message-summary")) {
    #	if (lookup("location")) {
    #		t_relay();
    #	};
    #	break;
    #};
# ------------------------------------------------------------------------
    # OPTIONS Section
    # ------------------------------------------------------------------------
    if (method=="OPTIONS") {
    	options_reply();
    	break;
    };
# ------------------------------------------------------------------------
    # NAT Test Section #1
    # ------------------------------------------------------------------------
    if (method=="REGISTER" && client_nat_test("3")) {
    	fix_contact();
    	force_rport();
    	setflag(2);
    };
# ------------------------------------------------------------------------
    # Click2Dial Section # 14
    # ------------------------------------------------------------------------
    #if (src_ip=="99.99.99.99" && (search("^From: <sip:700@") ||
search("^To: <sip:700@"))) {
    #	setflag(14);
    #};
# ------------------------------------------------------------------------
    # Registration Section
    # ------------------------------------------------------------------------
    if (method=="REGISTER") {
# allow all requests from user 700 - the Click2Dial controller
    	if (!isflagset(14)) {
    		if (!is_from_local()) {
    			sl_send_reply("403", "Unknown Domain");
    			break;
    		};
    		if (!www_authorize("", "subscriber")) {
    			www_challenge("", "0");
    			break;
    		};
    		if (!check_to()) {
    			sl_send_reply("401", "Unauthorized");
    			break;
    		};
    		
    		# To - Use To username and (optionally) domain to check
    		if (is_user_in("To", "demo-disabled")) {
    			sl_send_reply("403", "Your evaluation period has expired");
    			break;
    		};
    		
    		# To - Use To username and (optionally) domain to check
    		if (is_user_in("To", "disabled")) {
    			sl_send_reply("403", "Your account has been disabled");
    			break;
    		};
    	};
# snom sip phones use this header to start their 
    	# keep-alive mechanism for NAT bindings
    	append_to_reply("P-NAT-Refresh: 15\r\n");
if (!save("location")) {
    		sl_reply_error();
    	};
    	break;
    };
# ------------------------------------------------------------------------
    # Accounting Section # 1
    #
    # NOTE: We test for flag 14 because we do not want to record Click2Dial
    #       entries
    # ------------------------------------------------------------------------
    if ((method=="INVITE" || method=="BYE") && !isflagset(14)) {
    	setflag(1);
    };
# ------------------------------------------------------------------------
    # NAT Tear-Down Section
    # ------------------------------------------------------------------------
    if ((method == "BYE" || method == "CANCEL")) {
    	end_media_session();
    };
# ------------------------------------------------------------------------
    # Record Route Section
    #
    # 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
    # ------------------------------------------------------------------------
    if (!method=="REGISTER") {
    	record_route();
    };
# ------------------------------------------------------------------------
    # Loose Route Section
    #
    # Grant route routing if route headers present
    # ------------------------------------------------------------------------
    if (loose_route()) {
    	route(2);
    	break;
    };
# ------------------------------------------------------------------------
    # Prefix To Domain Section
    #
    # Convert the prefix defined on grp table to its own domain name
    # ------------------------------------------------------------------------
    prefix2domain();
# ------------------------------------------------------------------------
    # Message Waiting Indicator SUBSCRIBE Section
    #
    # Make sure all SUBSCRIBE messages with header "Event: message-summary"
    # are sent to the voicemail server
    # ------------------------------------------------------------------------
    if ((method=="SUBSCRIBE") && search("^Event: message-summary")) {
    	rewritehostport("xxx.xxx.190.243:5060");
    	t_relay();
    	break;
    };
# ------------------------------------------------------------------------
    # NAT Test Section #1
    # ------------------------------------------------------------------------
    if (client_nat_test("3") && !search("^Record-Route:")) {
    	force_rport();
    	fix_contact();
    };
# ------------------------------------------------------------------------
    # Enhanced Voicemail Applications Section
    #
    # Access additional Asterisk applications at extension 701 and 702
    # ------------------------------------------------------------------------
    #if (uri=~"^sip:70[1|2]@") {
    #	route(5);
    #	break;
    #};
# ------------------------------------------------------------------------
    # PSTN Section
    # ------------------------------------------------------------------------
    #
    # This is an example of US phone call.
    #
    #if (method=="INVITE") {
    #	if (uri=~"^sip:1[0-9]{10}@") {
    #		strip(1);
    #	};
    #	if ((uri=~"^sip:911@.*") || (uri=~"^sip:900[0-9]{7}@") ||
(uri=~"^sip:976[0-9]{7}@")) {
    #		sl_send_reply("503", "Service Unavailable");
    #		break;
    #	};
    #};
    if (method=="INVITE") {  ## Deny PSTN to 0204 and 095 on Tawan
    	if ((uri=~"^sip:0204[0-9]*@") || (uri=~"^sip:095[0-9]*@")) {
    		sl_send_reply("503", "Service Unavailable");
    		break;
    	};
    };
# ------------------------------------------------------------------------
    # Alias Routing Section
    # ------------------------------------------------------------------------
    lookup("aliases");
    if (!uri==myself) {
    	route(2);
    	break;
    };
# ------------------------------------------------------------------------
    # Load ACL Section
    # ------------------------------------------------------------------------
    if (method=="INVITE" && !isflagset(14)) {
    	xlog("L_ERR", "Time:[%Tf] Method:<%rm> r-uri:<%ru>\n");
    	xlog("L_ERR", "IP:<%is> From:<%fu> To:<%tu> %ct\n");
    	#if (is_user_in("Request-URI", "voicemail")) {
    	#	setflag(31);
    	#};
    	if (is_from_local() || is_uri_host_local()) {
    		#if (is_user_in("From", "411")) {
    		#	setflag(30);   ## some call from directory service
    		#};
    		
    		# Check user from grp table
    		if (is_user_in("From", "int")) {
    		  log(1, "SER: a INT user\n");
    			setflag(29);
    		};
    		if (is_user_in("From", "free-pstn")) {
    		  log(1, "SER: a FREE-PSTN user\n");
    			setflag(28);
    		};
    		if (avp_db_load("$from/username", "s:callidblock")) {
    			if (avp_check("s:callidblock", "eq/y/i")) {
    				setflag(25);
    			};
    		};
    	};
    	if (avp_db_load("$ruri/username", "s:anoncallrej")) {
    		if (avp_check("s:anoncallrej", "eq/y/i")) {
    		  log(1, "SER: a ANON-CALL-REJ user\n");
    			setflag(24);
    		};
    	};
    };
# ------------------------------------------------------------------------
    # Anonymous Call Rejection Section # 24
    # ------------------------------------------------------------------------
    if (isflagset(24) && (method=="INVITE") &&
search("^(f|F)rom:.*(a|A)nonymous")) {
    	route(8);
    	break;
    };
# ------------------------------------------------------------------------
    # 411 Directory Assistance Section # 30
    # ------------------------------------------------------------------------
    #if (method=="INVITE" && uri=~"^sip:411@.*") {
    #	if (isflagset(30)) {
    #		avp_write("$from", "i:34");
    #		avp_pushto("$ruri", "i:34");
    #		strip_tail(7);
    #		subst_uri('/^sip:([0-9]+)@(.*)$/sip:\15551212@\2/i');
    #		avp_delete("i:34");
    #		route(3);
    #		break;
    #	};
    #	sl_send_reply("503", "Service Unavailable");
    #	break;
    #};
# ------------------------------------------------------------------------
    # Speed Dialing Section
    # ------------------------------------------------------------------------
    if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) {
    	sd_lookup("speed_dial");
    };
# ------------------------------------------------------------------------
    # 011 International Call Section # 29
    # ------------------------------------------------------------------------
    #if (method=="INVITE" && uri=~"^sip:011[0-9]*@") {
    #	if (isflagset(29)) {
    #		route(6);
    #	} else {
    #		sl_send_reply("503", "Service Unavailable");
    #	};
    #	break;
    #};
# ------------------------------------------------------------------------
    # 002 International Call Section # 29
    # ------------------------------------------------------------------------
    if (method=="INVITE" && uri=~"^sip:002[0-9]*@") {
    	if (isflagset(29)) {
    		log(1, "SER: an International Call route(6)\n");
    		route(6);
    	} else {
    		sl_send_reply("503", "Service Unavailable");
    	};
    	break;
    };
# ------------------------------------------------------------------------
    # 0XXXXXXXXX Domestic Call Section # 29
    # ------------------------------------------------------------------------
    if (method=="INVITE" && uri=~"^sip:0[0-9]{9}@") {
    	if (isflagset(29)) {
    	  log(1, "SER: a Domestic Call route(6)\n");
    		route(6);
    	} else {
    		sl_send_reply("503", "Service Unavailable");
    	};
    	break;
    };
# ------------------------------------------------------------------------
    # URI Compare Section
    #
    # Here we compare the "from" and "to" to see if the caller is dialing
    # their own extension. If so then we route to voicemail(31) if needed
    # ------------------------------------------------------------------------
    if (method=="INVITE") {
    	avp_write("$from", "i:34");
    	if (avp_check("i:34", "eq/$ruri/i")) {
    		if (isflagset(31)) {
    			route(5);
    			break;
    		} else {
    			sl_send_reply("486", "Busy");
    			break;
    		}; 
    	};
    };
# ------------------------------------------------------------------------
    # Call Block Section
    # ------------------------------------------------------------------------
    #if (is_caller_blocked()) {
    #	route(7);
    #	break;
    #};
# ------------------------------------------------------------------------
    # Do Not Disturb Section
    # ------------------------------------------------------------------------
    if (avp_db_load("$ruri/username", "s:donotdisturb")) {
    	if (avp_check("s:donotdisturb", "eq/y/i")) {
    		route(5);
    		break;
    	};
    };
# ------------------------------------------------------------------------
    # Blind Call Forwarding Section
    # ------------------------------------------------------------------------
    if (method=="INVITE") {
      log(1, "SER: BLIND CALL FORWARDING\n");
    	# here we must store the current (aka original) R-URI because if
    	# we set call forwarding and the forwarded number is busy then we
    	# need to use this original R-URI to determine which voicemail
    	# box we should go to
    	if (isflagset(31)) {
    		avp_write("$ruri", "$voicemail");
    	};
    	if (avp_db_load("$ruri/username", "s:callfwd")) {
    		avp_pushto("$ruri", "s:callfwd");
    
    		# lookup the call fowarding number to see if it is a served
    		# sip number or a PSTN number
    
    		# check forwarding number rules
    		log(1, "SER: Check Forwarding Number Rules\n");
    		route(1);
    
    		#if (avp_check("$calltype", "eq/-/i")) {
    		#	log(1, "SER: 503 Service Unavailable 1\n");
    		#	sl_send_reply("503", "Service Unavailable");
    		#	break;
    		#};
    
    		# test for domestic PSTN gateway
    		if (uri=~"^sip:0[1-9]{9}@") {
    		#if (avp_check("$calltype", "eq/dom/i")) {
    		  log(1, "SER: Start a domestic PSTN call route(3)\n");
    			route(3);
    			break;
    		};
    
    		# test for international PSTN gateway
    		if (uri=~"^sip:002[1-9]*@") {
    		# if (avp_check("$calltype", "eq/int/i")) {
    			log(1, "SER: Start an international PSTN call route(6)\n");
    			route(6);
    			break;
    		};
    	};
    };
# ------------------------------------------------------------------------
    # Call Routing Section
    # ------------------------------------------------------------------------
    if (!lookup("location")) {
    
    	# if flag 31 (ie voicemail) is set and we made it here this means
    	# the user's phone is not registered anywhere. We'll forward to
    	# voicemail after this block because we need to check the call
    	# forward settings first
    	if (isflagset(31)) {
    		# flag 19 means the user has voicemail but is not online
    		# so we need to remember to send to voicemail if call
    		# forwarding is not enabled
    		setflag(19);
    	};
    
    	if (method=="INVITE") {
    		if (does_uri_exist()) {
    			# subscriber record found, but they're offline
    			log(1, "SER: Temporarily Unavailable\n");
    			sl_send_reply("480", "Temporarily Unavailable");
    			break;
    		};
    
    		if (uri=~"^sip:0[0-9]{9}@") {
    			# Send to PSTN Gateway
    			if (isflagset(28) || (is_user_in("From", "int")) ||
(is_user_in("From", "free-pstn"))) {
    			  log(1, "SER: Start a PSTN call\n");
    				route(3);
    			} else {
    				log(1, "SER: 503 Service Unavailable 2\n");
    				sl_send_reply("503", "Service Unavailable");
    			};
    			break;
    		};
    		sl_send_reply("404", "User Not Found");
    		break;
    	};
    };
# ------------------------------------------------------------------------
    # Call Forwarding Section
    # ------------------------------------------------------------------------
    if (method=="INVITE") {
    
    	# save R-URI in a temp AVP for later use
    	avp_write("$ruri", "i:99");
    
    	# only load the forward no answer option if voice mail is not enabled
    	if (!isflagset(31)) {
    		if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
    			route(1);
    			#if (!avp_check("$calltype", "eq/-/i")) {
    			#	if (avp_check("$calltype", "eq/dom/i")) {
    			#		avp_write("dom", "$fwd_no_answer_type");
    			#	} else if (avp_check("$calltype", "eq/int/i")) {
    			#		avp_write("int", "$fwd_no_answer_type");
    			#	} else {
    			#		avp_write("sip", "$fwd_no_answer_type");
    			#	}
    				setflag(27);
    			#};
    		};
    	};
    	if (avp_db_load("$ruri/username", "s:fwdbusy")) {
    		route(1);
    		#if (!avp_check("$calltype", "eq/-/i")) {
    		#	if (avp_check("$calltype", "eq/dom/i")) {
    		#		avp_write("dom", "$fwd_busy_type");
    		#	} else if (avp_check("$calltype", "eq/int/i")) {
    		#		avp_write("int", "$fwd_busy_type");
    		#	} else {
    		#		avp_write("sip", "$fwd_busy_type");
    		#	}
    			setflag(26);
    		#};
    	};
    	avp_pushto("$ruri", "i:99");
    };
    if (isflagset(19)) {
    	# send to voicemail
    	route(5);
    } else {
    	route(2);
    };
}
route[1] {
# Here we have route checks for all the call forwarding stuff. 
    # The return values are passed as AVP $calltype as follows:
    #
    # "-"   = R-URI is not allowed
    # "dom" = R-RURI is a domestic call
    # "int" = R-RURI is an international call
    # "sip" = R-RURI is a sip call
avp_write("sip", "$calltype");
# US international call
    #if (uri=~"^sip:1[0-9]{10}@") {
    #	strip(1);
    #};
# Call rejected                                                        
    #if ((uri=~"^sip:70[1|2]@") || (uri=~"^sip:411@.*") || 
    #    (uri=~"^sip:911@.*")   || (uri=~"^sip:900[0-9]{7}@") || 
    #    (uri=~"^sip:976[0-9]{7}@")) {
    #	break;
    #};
    if ((uri=~"^sip:0204[0-9]*@") || 
        (uri=~"^sip:095[0-9]*@") || 
        (uri=~"^sip:[0-9]{3}@")) {
    	break;
    };
  log(1, "SER: Look aliases\n");
    lookup("aliases");
    
    log(1, "SER: Look location\n");
    if (!lookup("location")) {
    	if (uri=~"^sip:0[0-9]{9}@") {
    		# test for domestic PSTN number
    		if (isflagset(28)) {
    			log(1, "SER isflagset 28 (domestic)\n");
    			avp_write("dom", "$calltype");
    		};
    	} else if (uri=~"^sip:002[0-9]*@") {
    		# test for international PSTN number
    		if (isflagset(29)) {
    			log(1, "SER isflagset 29 (int)\n");
    			avp_write("int", "$calltype");
    		};
    	};
    	break;
    };
    log(1, "SER isflagset (sip)\n");
    avp_write("sip", "$calltype");
}
route[2] {
    log(1, "SER: SIP Call On-Net section route(2)\n");
    if ((method=="INVITE") && !allow_trusted()) {
    	if (!proxy_authorize("", "subscriber")) {
    		proxy_challenge("", "0");
    		break;
    	} else if (!check_from()) {
    		log(1, "Spoofed SIP call attempt");
    		sl_send_reply("403", "Use From=ID");
    		break;
    	} else if (!(is_from_local() || is_uri_host_local())) {
    		sl_send_reply("403", "Please register to use our service");
    		break;
    	};
    };
    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(25)) {
    	replace("^From:(.*)>" , "From: "Anonymous"
sip:someone@anonymous.invalid");
    };
    if (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    t_on_failure("1");
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[3] {
    log(1, "SER: International Call Off-Net section route(3)\n");
# All Domestic Calls Go To CISCO 5300
    if (method=="INVITE") {
    	if (!proxy_authorize("", "subscriber")) {
    		proxy_challenge("", "0");
    		break;
    	} else if (!check_from()) {
    		log(1, "Spoofed SIP call attempt");
    		sl_send_reply("403", "Use From=ID");
    		break;
    	} else if (!(is_from_local() || is_uri_host_local())) {
    		sl_send_reply("403", "Please register to use our service");
    		break;
    	};
    	# enable caller id blocking for PSTN calls
    	if (isflagset(25)) {
    		append_rpid_hf();
    	};
    };
    # SIP->PSTN calls get 45 seconds to timeout
    log(1, "SER: Connecting to PSTN.....\n");
    avp_write("i:45", "inv_timeout");
    rewritehost("xxx.xxx.190.243");
    #rewritehostport("xxx.xxx.190.248:5065");
    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 (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    if (isflagset(31)) {  # is voice mail?
    	t_on_failure("1");
    };
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[4] {
    log(1, "SER: Voice Mail section route(4)\n");
# voicemail route #1
    #
    # this path this executed during these conditions:
    #
    # cond 1) the called number is in the location table
    #		      but the callee did not answer the phone
    #		      (ie, failover to voicemail)
if (isflagset(25)) {
    	replace("^From:(.*)>" , "From: "Anonymous"
sip:someone@anonymous.invalid");
    };
    rewritehostport("99.99.99.100:5060");
    append_branch();
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[5] {
    log(1, "SER: Voice Mail section route(5)\n");
# voicemail route #2
    #
    # this path this executed during these conditions:
    #
    #	cond 1) the called number is not in the location table
    #	cond 2) the from_uri == to_uri (ie, caller==callee)
if (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    rewritehostport("99.999.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[6] {
    log(1, "SER: Domestic Call Off-Net section route(6)\n");
    
    # All International Calls Go To CISCO 5300
    if (method=="INVITE") {
    	if (!proxy_authorize("", "subscriber")) {
    		proxy_challenge("", "0");
    		break;
    	} else if (!check_from()) {
    		log(1, "Spoofed SIP call attempt");
    		sl_send_reply("403", "Use From=ID");
    		break;
    	} else if (!(is_from_local() || is_uri_host_local())) {
    		sl_send_reply("403", "Please register to use our service");
    		break;
    	};
    	# enable caller id blocking for PSTN calls
    	if (isflagset(25)) {
    		append_rpid_hf();
    	};
    };
    # SIP->PSTN calls get 45 seconds to timeout
    avp_write("i:45", "inv_timeout");
    rewritehost("xxx.xxx.190.243");
    #rewritehostport("xxx.xxx.190.248:5065");
    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 (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[7] {
  log(1, "SER: Caller Blocked section route(7)\n");
# caller blocked announcment
    #
    # this path this executed if a caller has been blocked
    
    if (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    rewriteuri("sip:699@99.99.99.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
route[8] {
    log(1, "SER: Anonymous Call Rejection section route(8)\n");
    
    # anonymous call rejection announcment
    #
    # this path this executed for anonymous callers
if (method=="INVITE" || method=="ACK") {
    	use_media_proxy();
    };
    rewriteuri("sip:698@99.99.99.100:5060");
    t_on_reply("1");
    if (!t_relay()) {
    	if (method=="INVITE" || method=="ACK") {
    		end_media_session();
    	};
    	sl_reply_error();
    };
}
onreply_route[1] {
    # Not all 2xx messages have a content body so here we
    # make sure our Content-Length > 0 to avoid a parse error
    if (status=~"(180)|(183)|2[0-9][0-9]") {
    	if (!search("^Content-Length:\ 0")) {
    		use_media_proxy();
    	};
    };
    if (client_nat_test("1")) {
    	fix_contact();
    };
}
failure_route[1] {
    log(1, "SER: Failure Route section failure_route(1)\n");
# if caller hung up then don't sent to voicemail
    if (t_check_status("487")) {
    	break;
    };
    if (isflagset(26) && t_check_status("486")) {
    	# forward busy is flag 26
    	
    	if (avp_pushto("$ruri", "s:fwdbusy")) {
    		log(1, "SER: fork to fwdbusy\n");
    		avp_delete("s:fwdbusy");
    		append_branch();
    		resetflag(26);
    		
    		# test for domestic PSTN gateway
    		if (uri=~"^sip:0[0-9]{9}@") {
    		# if (avp_check("$fwd_busy_type", "eq/dom/i")) {
    			# test for domestic PSTN gateway
    			log(1, "SER: Busy Failure and Jump to route(6)\n");
    			route(6);
    		} else if (uri=~"^sip:002[1-9][0-9]*@") {
    		# } else if (avp_check("$fwd_busy_type", "eq/int/i")) {
    			# test for international PSTN gateway
    			log(1, "SER: Busy Failure and Jump to route(3)\n");
    			route(3);
    		} else {
    			# default to sip call
    			log(1, "SER: Busy Failure and Jump to route(2)\n");
    			route(2);
    		};
    		break;
    	};
    };
# here we can have either voicemail __OR__ forward no answer
    if (isflagset(27) && t_check_status("408")) {
    	# forward no answer is flag 27
    	
    	if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
    		avp_pushto("$ruri", "s:fwdnoanswer");
    		log(1, "SER: fork to fwdnoanswer\n");
    		avp_delete("s:fwdnoanswer");
    		append_branch();
    		resetflag(27);
    		
    		if (uri=~"^sip:0[0-9]{9}@") {
    		# if (avp_check("$fwd_no_answer_type", "eq/dom/i")) {
    			# test for domestic PSTN gateway
    			log(1, "SER: No Answer Failure and Jump to route(3)\n");
    			route(3);
    	  } else if (uri=~"^sip:002[1-9][0-9]*@") {
    		# } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) {
    			# test for international PSTN gateway
    			log(1, "SER: No Answer Failure and Jump to route(6)\n");
    			route(6);
    		} else {
    			# default to sip call
    			log(1, "SER: No Answer Failure and Jump to route(2)\n");				
    			route(2);
    		};
    		break;
    	};
    } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) {
    	avp_delete("$voicemail");
    	log(1, "SER: No Answer Failure and Jump to route(4)\n");
    	route(4);
    	break;
    };
}

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Re: [Serusers] Help when make a Call Busy Forward Failed? Does 'bestpractice' release??