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

[Greger V. Teigre]

Sorry, Charles, but your ser.cfg is too unwieldy (and with no indentation) 
to successfully be able to trace the routing logic.  However, trying to go 
through the logic in my head, it seems that there is something wrong with 
your basic logic.  Remember: every single message will start at the top 
route{} and then go through your logic until reaching a t_relay (or 
similar), sl_send_reply or break.  If you don't call t_relay, the route[x] 
will return to the place where it was called and then continue processing. 
Your use of flags to control the routing makes ser.cfg very hard to read.
    My suggestion is that you strip away everything you don't need just to 
test what you are trying to achieve.  When posting something, include your 
simplified ser.cfg, as well as an ngrep trace, it is a lot easier to help 
you.
g-)

Charles Wang wrote:
> 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 at .*") || (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 at .*") { # 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 at .*") ||
> #    (uri=~"^sip:911 at .*")   || (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 at 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 at 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 at 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 at 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;
> };
> }