sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] Questions on using ser.cfg version 0.9.0 for call forward/busy/noanswer

by Charles Wang

Dear ALL: I think it is a very good chance to study Paul's ser.cfg file. I use the 0.9.0 sample cfg and try to implement a call forward, busy forward, no answer forward function. But I got a 503 message when I use make a call from UA 1011 to UA 1022(registed on the same ser) with callforward to a PSTN . But I have no voicemail option supported. So I must modify it. In usr_preferences table: username domain attribute value ------------------------------------------------------------------------------------------------------------------- 1022 ser.xxx.net.tw donotdisturb n 1022 ser.xxx.net.tw voicemail n 1022 ser.xxx.net.tw anoncallrej n 1022 ser.xxx.net.tw callidblock n 1022 ser.xxx.net.tw callfwd 0939749xxx(a)ser.xxx.net.tw 1022 ser.xxx.net.tw fwdnoanswer 0939749xxx(a)ser.xxx.net.tw 1022 ser.xxx.net.tw fwdbusy 0939749xxx(a)ser.xxx.net.tw In grp table: username domain grp --------------------------------------------------- 1022 ser.xxx.net.tw int 1011 ser.xxx.net.tw int I don't know where to defind the 'dom' and 'int' or '-' setting at DB table. What is its table name? domain ==> ser.xxx.net.tw (the same as caller and machine) When I test this call forward, a UA 1011 call to 1022. It It should be forward to a PSTN and use route(3) or route(6). But when I do it, it seems failed at lookup("aliases") and lookup("location") function. Log list below: --------------------------------------------------------------------------------------------------------- ser /usr/local/sbin/ser[9232]: SER: BLIND CALL FORWARDING ser /usr/local/sbin/ser[9232]: SER: Check Forwarding Number Rules ser /usr/local/sbin/ser[9232]: SER: Look aliases ser /usr/local/sbin/ser[9232]: ERROR: parse_uri: bad uri, state 0 parsed: <0939> (4) / <0939749xxx(a)ser.xxx.net.tw> (27) ser /usr/local/sbin/ser[9232]: extract_aor(): Error while parsing Address of Record ser /usr/local/sbin/ser[9232]: lookup(): Error while extracting address of record ser /usr/local/sbin/ser[9232]: SER: Look location ser /usr/local/sbin/ser[9232]: ERROR: parse_uri: bad uri, state 0 parsed: <0939> (4) / <0939749xxx(a)ser.xxx.net.tw> (27) ser /usr/local/sbin/ser[9232]: extract_aor(): Error while parsing Address of Record ser /usr/local/sbin/ser[9232]: lookup(): Error while extracting address of record ser /usr/local/sbin/ser[9232]: SER: 503 Service Unavailable --------------------------------------------------------------------------------------------------------- If I delete the record callfwd of 1022 and want to test call noanswer forward. It seems can not forward to a PSTN number and jump to route(3) or route(6). It always keep the orginial rule and use route(2). Also the error message "error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy" always displays at log file whether the call is connected or failed call. Log list below: ---------------------------------------------------------------------------------------------------------ser /usr/local/sbin/ser[9233]: SER: BLIND CALL FORWARDING ser /usr/local/sbin/ser[9233]: SER: Look aliases ser /usr/local/sbin/ser[9233]: SER: Look location ser /usr/local/sbin/ser[9233]: SER isflagset (sip) ser /usr/local/sbin/ser[9233]: SER: Look aliases ser /usr/local/sbin/ser[9233]: SER: Look location ser /usr/local/sbin/ser[9233]: SER isflagset (sip) ser /usr/local/sbin/ser[9233]: SER: SIP Call On-Net section route(2) ser /usr/local/sbin/ser[9233]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy ser /usr/local/sbin/ser[9231]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy ser /usr/local/sbin/ser[9231]: SER: SIP Call On-Net section route(2) ser /usr/local/sbin/ser[9234]: SER: Failure Route section failure_route(1) ser /usr/local/sbin/ser[9234]: SER: fork to fwdnoanswer ser /usr/local/sbin/ser[9234]: SER: No Answer Failure and Jump to route(2) ser /usr/local/sbin/ser[9234]: SER: SIP Call On-Net section route(2) ser /usr/local/sbin/ser[9234]: ERROR: parse_uri: bad uri, state 0 parsed: <0939> (4) / <0939749xxx(a)ser.xxx.net.tw> (27) ser /usr/local/sbin/ser[9234]: ERROR: parse_sip_msg_uri: bad uri <0939749xxx(a)ser.xxx.net.tw> ser /usr/local/sbin/ser[9234]: error: mediaproxy/getDestinationDomain(): error parsing destination URI ser /usr/local/sbin/ser[9234]: ERROR: parse_uri: bad uri, state 0 parsed: <0939> (4) / <0939749xxx(a)ser.xxx.net.tw> (27) ser /usr/local/sbin/ser[9234]: ERROR: parse_sip_msg_uri: bad uri <0939749xxx(a)ser.xxx.net.tw> ser /usr/local/sbin/ser[9234]: is_uri_host_local(): Error while parsing URI ser /usr/local/sbin/ser[9234]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy ser /usr/local/sbin/ser[9234]: ERROR: parse_uri: bad uri, state 0 parsed: <0939> (4) / <0939749xxx(a)ser.xxx.net.tw> (27) ser /usr/local/sbin/ser[9234]: ERROR: uri2proxy: bad_uri: 0939749xxx(a)ser.xxx.net.tw ser /usr/local/sbin/ser[9234]: ERROR: t_forward_nonack: failure to add branches ser /usr/local/sbin/ser[9234]: ERROR: w_t_relay (failure mode): forwarding failed ser /usr/local/sbin/ser[9234]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy ser /usr/local/sbin/ser[9234]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) ser /usr/local/sbin/ser[9231]: SER: SIP Call On-Net section route(2) --------------------------------------------------------------------------------------------------------- My ser.cfg list below: --------------------------------------------------------------------------------------------------------- # # $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=99.99.99.99 alias=ser.xxx.net.tw alias=ser alias=99.99.99.99 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" # ------------------ 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") # ------------------------- 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; }; # ------------------------------------------------------------------------ # 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; }; # ------------------------------------------------------------------------ # 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("99.99.99.100:5060"); # 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 of Taiwan 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("$calltype", "eq/-/i")) { log(1, "SER: 503 Service Unavailable\n"); sl_send_reply("503", "Service Unavailable"); break; }; # test for domestic PSTN gateway if (avp_check("$calltype", "eq/dom/i")) { log(1, "SER: Start a domestic PSTN call route(3)\n"); route(3); break; }; # test for international PSTN gateway if (avp_check("$calltype", "eq/int/i")) { log(1, "SER: Start an international PSTN call route(6)\n"); route(6); break; }; }; }; # ------------------------------------------------------------------------ # Call Routing Section # ------------------------------------------------------------------------ if (!lookup("location")) { # if flag 31 (ie voicemail) is set and we made it here this means # the user's phone is not registered anywhere. We'll forward to # voicemail after this block because we need to check the call # forward settings first if (isflagset(31)) { # flag 19 means the user has voicemail but is not online # so we need to remember to send to voicemail if call # forwarding is not enabled setflag(19); }; if (method=="INVITE") { if (does_uri_exist()) { # subscriber record found, but they're offline log(1, "SER: Temporarily Unavailable\n"); sl_send_reply("480", "Temporarily Unavailable"); break; }; if (uri=~"^sip:0[0-9]{9}@") { # Send to PSTN Gateway if (isflagset(28)) { log(1, "SER: Start a PSTN call\n"); route(3); } else { log(1, "SER: 503 Service Unavailable\n"); sl_send_reply("503", "Service Unavailable"); }; break; }; sl_send_reply("404", "User Not Found"); break; }; }; # ------------------------------------------------------------------------ # Call Forwarding Section # ------------------------------------------------------------------------ if (method=="INVITE") { # save R-URI in a temp AVP for later use avp_write("$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 $calltype as follows: # # "-" = R-URI is not allowed # "dom" = R-RURI is a domestic call # "int" = R-RURI is an international call # "sip" = R-RURI is a sip call avp_write("-", "$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; }; # Error Message from this <================== ERROR 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", "$calltype"); }; # } else if (uri=~"^sip:011[0-9]*@") { } else if (uri=~"^sip:002[0-9]*@") { # test for international PSTN number if (isflagset(29)) { log(1, "SER isflagset 29 (int)\n"); avp_write("int", "$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: Demestic 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 avp_write("i:45", "inv_timeout"); rewritehost("99.99.99.250"); 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)) { 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("99.99.99.100:5060"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[6] { log(1, "SER: International 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("99.99.99.240"); 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 (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: Busy Failure and Jump to route(3)\n"); route(3); } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: Busy Failure and Jump to route(6)\n"); route(6); } 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_pushto("$ruri", "s:fwdnoanswer")) { log(1, "SER: fork to fwdnoanswer\n"); avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (avp_check("$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 (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; }; }

19 years, 9 months

[Serusers] Juha: LCR Module Change

by Java Rockx

Juha, I have two gateways defined in the gw table and a have a few LCR rules in the lcr table. The one lcr entry is a "catch-all" which would be used if all other PSTN gateways failed. My LCR table looks like this 'prefix','from_uri','grp_id','priority' '321939','%321251%','1','2' '407319','%321251%','1','2' '%','%','2','1' So if I dial from 3212518888 to 4073195555 then I would think at the gateway in group #1 would be used first and if it failed then the gateway in group #2 would be used. However when next_gw() is called the following SQL is executed: SELECT gw.ip_addr, gw.port FROM gw, lcr WHERE 'sip:3212518888@sip.mycompany.com' LIKE lcr.from_uri AND '4073195555' LIKE CONCAT(lcr.prefix, '%') AND lcr.grp_id = gw.grp_id ORDER BY CHAR_LENGTH(lcr.prefix), lcr.priority DESC, RAND() The ORDER BY clause causes the group #2 gateway to appear first in the result set because '%' sorts before '407319'. So should this SQL ORDER BY clause be changed to ORDER BY CHAR_LENGTH(lcr.prefix) DESC, lcr.priority DESC, RAND() or is there a better way to specify a "catch-all" gateway? Regards, Paul

19 years, 9 months

[Serusers] SER+Asterisk+NAT. Need a working example or need to talk about it.

by Alexey N. Kovyrin ＠ Home

Hello! I'm implementing some VoIP system with SER+Asterisk. I've already implemented many features, but all of them was tested only in non-NAT environments. Now, at final-alpha stage (before starting massive beta tests) I found big (as for me) problem with NAT... Now I need a working example of ser.cfg (it will be really good) or, simply, to talk with someone, who realized such scheme... My current problem is with ua-to-ua calls between my users... I have a following call routing scheme... UA1->...->SER->Asterisk->SER->...->UA2... Description: UA1 calls to some number, ser routes this call to asterisk, asterisk resolves that number to some account (login name, that being used by users to login to ser) and makes call to sip/ser/user_name... and ser routes this call for user with such login... I have strange problems with natted UAs... Please, send me a working example of similiar schemes (ser+asterisk+nat)... or help me by a word ;-) Great thanks! -- /Scoundrel

19 years, 9 months

[Serusers] Cannot login from remote site

by ST Ooi

Hi, I´m new to SER and trying to login from remote site using Xten Lite software, however it gave me login timeout error, below are 3 of the packets that I capture using ethereal, can anyone please help? BTW, I installed the SER on a PIV 1.6 PC. Fedora Core 3. Firewall disabled. No. Time Source Destination Protocol Info 59 40.152648 210.186.103.104 192.168.123.123 SIP Request: REGISTER sip:sipserver.shop-streets.com Frame 59 (534 bytes on wire, 534 bytes captured) Ethernet II, Src: 00:05:25:40:2e:1d, Dst: 00:c0:26:84:8a:44 Internet Protocol, Src Addr: 210.186.103.104 (210.186.103.104), Dst Addr: 192.168.123.123 (192.168.123.123) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Session Initiation Protocol (SIP as raw text) 0000 00 c0 26 84 8a 44 00 05 25 40 2e 1d 08 00 45 00 ..&..D..%@....E. 0010 02 08 58 cd 00 00 79 11 70 d1 d2 ba 67 68 c0 a8 ..X...y.p...gh.. 0020 7b 7b 13 c4 13 c4 01 f4 1e c8 52 45 47 49 53 54 {{........REGIST 0030 45 52 20 73 69 70 3a 73 69 70 73 65 72 76 65 72 ER sip:sipserver 0040 2e 73 68 6f 70 2d 73 74 72 65 65 74 73 2e 63 6f .shop-streets.co 0050 6d 20 53 49 50 2f 32 2e 30 0d 0a 56 69 61 3a 20 m SIP/2.0..Via: 0060 53 49 50 2f 32 2e 30 2f 55 44 50 20 32 31 30 2e SIP/2.0/UDP 210. 0070 31 38 36 2e 31 30 33 2e 31 30 34 3a 35 30 36 30 186.103.104:5060 0080 3b 72 70 6f 72 74 3b 62 72 61 6e 63 68 3d 7a 39 ;rport;branch=z9 0090 68 47 34 62 4b 42 35 43 37 39 45 31 34 43 38 45 hG4bKB5C79E14C8E 00a0 46 34 41 30 36 39 33 31 36 39 38 45 35 30 33 37 F4A06931698E5037 00b0 45 31 37 39 32 0d 0a 46 72 6f 6d 3a 20 53 54 20 E1792..From: ST 00c0 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 40 73 Ooi <sip:88887@s 00d0 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 ipserver.shop-st 00e0 72 65 65 74 73 2e 63 6f 6d 3e 3b 74 61 67 3d 33 reets.com>;tag=3 00f0 37 37 32 36 35 34 31 34 34 0d 0a 54 6f 3a 20 53 772654144..To: S 0100 54 20 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 T Ooi <sip:88887 0110 40 73 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d @sipserver.shop- 0120 73 74 72 65 65 74 73 2e 63 6f 6d 3e 0d 0a 43 6f streets.com>..Co 0130 6e 74 61 63 74 3a 20 22 53 54 20 4f 6f 69 22 20 ntact: "ST Ooi" 0140 3c 73 69 70 3a 38 38 38 38 37 40 32 31 30 2e 31 <sip:88887@210.1 0150 38 36 2e 31 30 33 2e 31 30 34 3a 35 30 36 30 3e 86.103.104:5060> 0160 0d 0a 43 61 6c 6c 2d 49 44 3a 20 39 45 36 34 35 ..Call-ID: 9E645 0170 42 45 41 31 44 35 45 34 32 31 34 41 35 32 34 36 BEA1D5E4214A5246 0180 36 32 44 44 42 33 45 39 36 35 32 40 73 69 70 73 62DDB3E9652@sips 0190 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 72 65 65 erver.shop-stree 01a0 74 73 2e 63 6f 6d 0d 0a 43 53 65 71 3a 20 31 34 ts.com..CSeq: 14 01b0 39 38 35 20 52 45 47 49 53 54 45 52 0d 0a 45 78 985 REGISTER..Ex 01c0 70 69 72 65 73 3a 20 31 38 30 30 0d 0a 4d 61 78 pires: 1800..Max 01d0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 55 -Forwards: 70..U 01e0 73 65 72 2d 41 67 65 6e 74 3a 20 58 2d 4c 69 74 ser-Agent: X-Lit 01f0 65 20 72 65 6c 65 61 73 65 20 31 31 30 33 6d 0d e release 1103m. 0200 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a .Content-Length: 0210 20 30 0d 0a 0d 0a 0.... No. Time Source Destination Protocol Info 64 40.222196 192.168.123.123 sipserver.shop-streets.com SIP Request: REGISTER sip:sipserver.shop-streets.com Frame 64 (662 bytes on wire, 662 bytes captured) Ethernet II, Src: 00:c0:26:84:8a:44, Dst: 00:05:25:40:2e:1d Internet Protocol, Src Addr: 192.168.123.123 (192.168.123.123), Dst Addr: sipserver.shop-streets.com (218.111.70.155) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Session Initiation Protocol (SIP as raw text) 0000 00 05 25 40 2e 1d 00 c0 26 84 8a 44 08 00 45 10 ..%@....&..D..E. 0010 02 88 01 2a 40 00 40 11 d9 fc c0 a8 7b 7b da 6f ...*@.@.....{{.o 0020 46 9b 13 c4 13 c4 02 74 5f b4 52 45 47 49 53 54 F......t_.REGIST 0030 45 52 20 73 69 70 3a 73 69 70 73 65 72 76 65 72 ER sip:sipserver 0040 2e 73 68 6f 70 2d 73 74 72 65 65 74 73 2e 63 6f .shop-streets.co 0050 6d 20 53 49 50 2f 32 2e 30 0d 0a 52 65 63 6f 72 m SIP/2.0..Recor 0060 64 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a 31 39 d-Route: <sip:19 0070 32 2e 31 36 38 2e 31 32 33 2e 31 32 33 3b 66 74 2.168.123.123;ft 0080 61 67 3d 33 37 37 32 36 35 34 31 34 34 3b 6c 72 ag=3772654144;lr 0090 3d 6f 6e 3e 0d 0a 56 69 61 3a 20 53 49 50 2f 32 =on>..Via: SIP/2 00a0 2e 30 2f 55 44 50 20 31 39 32 2e 31 36 38 2e 31 .0/UDP 192.168.1 00b0 32 33 2e 31 32 33 3b 62 72 61 6e 63 68 3d 7a 39 23.123;branch=z9 00c0 68 47 34 62 4b 30 38 38 39 2e 34 34 65 65 33 62 hG4bK0889.44ee3b 00d0 35 37 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 57.0..Via: SIP/2 00e0 2e 30 2f 55 44 50 20 32 31 30 2e 31 38 36 2e 31 .0/UDP 210.186.1 00f0 30 33 2e 31 30 34 3a 35 30 36 30 3b 72 70 6f 72 03.104:5060;rpor 0100 74 3d 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 t=5060;branch=z9 0110 68 47 34 62 4b 42 35 43 37 39 45 31 34 43 38 45 hG4bKB5C79E14C8E 0120 46 34 41 30 36 39 33 31 36 39 38 45 35 30 33 37 F4A06931698E5037 0130 45 31 37 39 32 0d 0a 46 72 6f 6d 3a 20 53 54 20 E1792..From: ST 0140 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 40 73 Ooi <sip:88887@s 0150 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 ipserver.shop-st 0160 72 65 65 74 73 2e 63 6f 6d 3e 3b 74 61 67 3d 33 reets.com>;tag=3 0170 37 37 32 36 35 34 31 34 34 0d 0a 54 6f 3a 20 53 772654144..To: S 0180 54 20 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 T Ooi <sip:88887 0190 40 73 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d @sipserver.shop- 01a0 73 74 72 65 65 74 73 2e 63 6f 6d 3e 0d 0a 43 6f streets.com>..Co 01b0 6e 74 61 63 74 3a 20 22 53 54 20 4f 6f 69 22 20 ntact: "ST Ooi" 01c0 3c 73 69 70 3a 38 38 38 38 37 40 32 31 30 2e 31 <sip:88887@210.1 01d0 38 36 2e 31 30 33 2e 31 30 34 3a 35 30 36 30 3e 86.103.104:5060> 01e0 0d 0a 43 61 6c 6c 2d 49 44 3a 20 39 45 36 34 35 ..Call-ID: 9E645 01f0 42 45 41 31 44 35 45 34 32 31 34 41 35 32 34 36 BEA1D5E4214A5246 0200 36 32 44 44 42 33 45 39 36 35 32 40 73 69 70 73 62DDB3E9652@sips 0210 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 72 65 65 erver.shop-stree 0220 74 73 2e 63 6f 6d 0d 0a 43 53 65 71 3a 20 31 34 ts.com..CSeq: 14 0230 39 38 35 20 52 45 47 49 53 54 45 52 0d 0a 45 78 985 REGISTER..Ex 0240 70 69 72 65 73 3a 20 31 38 30 30 0d 0a 4d 61 78 pires: 1800..Max 0250 2d 46 6f 72 77 61 72 64 73 3a 20 36 39 0d 0a 55 -Forwards: 69..U 0260 73 65 72 2d 41 67 65 6e 74 3a 20 58 2d 4c 69 74 ser-Agent: X-Lit 0270 65 20 72 65 6c 65 61 73 65 20 31 31 30 33 6d 0d e release 1103m. 0280 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a .Content-Length: 0290 20 30 0d 0a 0d 0a 0.... No. Time Source Destination Protocol Info 65 40.312671 192.168.123.123 sipserver.shop-streets.com SIP Request: REGISTER sip:sipserver.shop-streets.com Frame 65 (662 bytes on wire, 662 bytes captured) Ethernet II, Src: 00:c0:26:84:8a:44, Dst: 00:05:25:40:2e:1d Internet Protocol, Src Addr: 192.168.123.123 (192.168.123.123), Dst Addr: sipserver.shop-streets.com (218.111.70.155) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Session Initiation Protocol (SIP as raw text) 0000 00 05 25 40 2e 1d 00 c0 26 84 8a 44 08 00 45 10 ..%@....&..D..E. 0010 02 88 01 2b 40 00 40 11 d9 fb c0 a8 7b 7b da 6f ...+@.@.....{{.o 0020 46 9b 13 c4 13 c4 02 74 5f b4 52 45 47 49 53 54 F......t_.REGIST 0030 45 52 20 73 69 70 3a 73 69 70 73 65 72 76 65 72 ER sip:sipserver 0040 2e 73 68 6f 70 2d 73 74 72 65 65 74 73 2e 63 6f .shop-streets.co 0050 6d 20 53 49 50 2f 32 2e 30 0d 0a 52 65 63 6f 72 m SIP/2.0..Recor 0060 64 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a 31 39 d-Route: <sip:19 0070 32 2e 31 36 38 2e 31 32 33 2e 31 32 33 3b 66 74 2.168.123.123;ft 0080 61 67 3d 33 37 37 32 36 35 34 31 34 34 3b 6c 72 ag=3772654144;lr 0090 3d 6f 6e 3e 0d 0a 56 69 61 3a 20 53 49 50 2f 32 =on>..Via: SIP/2 00a0 2e 30 2f 55 44 50 20 31 39 32 2e 31 36 38 2e 31 .0/UDP 192.168.1 00b0 32 33 2e 31 32 33 3b 62 72 61 6e 63 68 3d 7a 39 23.123;branch=z9 00c0 68 47 34 62 4b 30 38 38 39 2e 34 34 65 65 33 62 hG4bK0889.44ee3b 00d0 35 37 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 57.0..Via: SIP/2 00e0 2e 30 2f 55 44 50 20 32 31 30 2e 31 38 36 2e 31 .0/UDP 210.186.1 00f0 30 33 2e 31 30 34 3a 35 30 36 30 3b 72 70 6f 72 03.104:5060;rpor 0100 74 3d 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 t=5060;branch=z9 0110 68 47 34 62 4b 42 35 43 37 39 45 31 34 43 38 45 hG4bKB5C79E14C8E 0120 46 34 41 30 36 39 33 31 36 39 38 45 35 30 33 37 F4A06931698E5037 0130 45 31 37 39 32 0d 0a 46 72 6f 6d 3a 20 53 54 20 E1792..From: ST 0140 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 40 73 Ooi <sip:88887@s 0150 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 ipserver.shop-st 0160 72 65 65 74 73 2e 63 6f 6d 3e 3b 74 61 67 3d 33 reets.com>;tag=3 0170 37 37 32 36 35 34 31 34 34 0d 0a 54 6f 3a 20 53 772654144..To: S 0180 54 20 4f 6f 69 20 3c 73 69 70 3a 38 38 38 38 37 T Ooi <sip:88887 0190 40 73 69 70 73 65 72 76 65 72 2e 73 68 6f 70 2d @sipserver.shop- 01a0 73 74 72 65 65 74 73 2e 63 6f 6d 3e 0d 0a 43 6f streets.com>..Co 01b0 6e 74 61 63 74 3a 20 22 53 54 20 4f 6f 69 22 20 ntact: "ST Ooi" 01c0 3c 73 69 70 3a 38 38 38 38 37 40 32 31 30 2e 31 <sip:88887@210.1 01d0 38 36 2e 31 30 33 2e 31 30 34 3a 35 30 36 30 3e 86.103.104:5060> 01e0 0d 0a 43 61 6c 6c 2d 49 44 3a 20 39 45 36 34 35 ..Call-ID: 9E645 01f0 42 45 41 31 44 35 45 34 32 31 34 41 35 32 34 36 BEA1D5E4214A5246 0200 36 32 44 44 42 33 45 39 36 35 32 40 73 69 70 73 62DDB3E9652@sips 0210 65 72 76 65 72 2e 73 68 6f 70 2d 73 74 72 65 65 erver.shop-stree 0220 74 73 2e 63 6f 6d 0d 0a 43 53 65 71 3a 20 31 34 ts.com..CSeq: 14 0230 39 38 35 20 52 45 47 49 53 54 45 52 0d 0a 45 78 985 REGISTER..Ex 0240 70 69 72 65 73 3a 20 31 38 30 30 0d 0a 4d 61 78 pires: 1800..Max 0250 2d 46 6f 72 77 61 72 64 73 3a 20 36 39 0d 0a 55 -Forwards: 69..U 0260 73 65 72 2d 41 67 65 6e 74 3a 20 58 2d 4c 69 74 ser-Agent: X-Lit 0270 65 20 72 65 6c 65 61 73 65 20 31 31 30 33 6d 0d e release 1103m. 0280 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a .Content-Length: 0290 20 30 0d 0a 0d 0a 0.... Thanks in advance ST Ooi ________________________________________________ Message sent using UebiMiau 2.7.2 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 2/18/2005

19 years, 9 months

[Serusers] Warning: sl_send_reply: I won't send a reply for ACK!!

by Java Rockx

Hi all. How can I prevent this warning from occuring? Warning: sl_send_reply: I won't send a reply for ACK!! Regards, Paul

19 years, 9 months

Re: [Serusers] How to use avp_aliases?

by Greger V. Teigre

Basically what you do is to load the number to forward to by using avp_db_load and then add code in failure_route to test on the return codes, write the loaded number into ruri and use append_branch() to fork a new request. Paul posted an example on serusers http://lists.iptel.org/pipermail/serusers/2005-January/014807.html He uses a more complex authorization structure for giving users access to various types of calls (local, long distance, international), so you will have to sort out the code necessary for the forwarding you want to do. Here is what you need on failure_route[1] (remember to set t_on_failure[1] before calling t_relay() ): # flag 26 means that a "forward busy" number is set # This flag is set if avp_db_load("$ruri","s:fwdbusy") resulted in a an avpair being loaded (before t_relay()) # Code example using only username portion of ruri to look up avpair: if (avp_db_load("$ruri/username", "s:fwdbusy")) { setflag(26) } if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # route the call to IP or PSTN route(2); break; }; }; g-) Charles Wang wrote: > Thank you for your kind to answer me these question. > And because I can't find a lot of discussions at serusers's maillist. > So I post it to serdev. I am very sorry to do that. > > I have no voicemail solution in my scenario. But I wanna implement the > call busy forward and noanswer forward to another UA or PSTN. > For example: if A call B and B is busy or noanswer, then forward this > call to C or PSTN(defined by B). Is it possible? > > Why does almost discussions about avpops are about voicemail? > Is it mean that system will send a voicemail to > "1011(a)domainA.com"(setting above at usr_preferences) if a UA is busy > or noanswer? > > How can I implement it if I want to forward the busy call or noanswer > call to PSTN or another UA(registed on the same machine)? > > > On Sat, 19 Feb 2005 12:02:18 +0100, Greger V. Teigre > <greger(a)teigre.com> wrote: >> I don't think your message is appropriate for serdev, so I removed >> it from the recipient list. >> See inline comments. >> >> Charles Wang wrote: >>> Dear ALL: >>> >>> I have defined these lines at my ser.cfg. >>> >>> modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") >>> modparam("avpops", "avp_table", "usr_preferences") >>> 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") >>> >>> Q1: If I use usr_preferences as default table, should I insert some >>> initial data to it? >>> Can you give me some example? >> >> Yes, you should. Let's say you want user a(a)b.com to have voicemail >> as an option. Add: >> >> username=a >> domain=b.com >> attribute=voicemail >> value=y >> type=0 (default string) >> >>> >>> And I have refered to http://www.voice-system.ro/docs/avpops >>> >>> And find many examples in it. >>> >>> But I still can't understand what it mean about avp_db_load(). >> >> avp_db_load() will load a certain value from your usr_preferences >> table. >> >> avp_db_load("$ruri","s:voicemail"); >> will use to request uri (i.e. somebody is calling a(a)b.com) to lookup >> the attribute voicemail. Instead of "$ruri", you can use >> "$ruri/username" or "$ruri/domain" and only the username or domain >> part of $ruri, respectively, will be used to look up the attribute >> voicemail in the table. avp_check("s:voicemail", "eq/y/i") can then >> be used to check if value from the usr_preferences table is "y". >> >>> my avp table is named as "usr_preferences" and it is empty before I >>> use avpops modules. >>> >>> For this web page said: >>> avp_db_load("$ruri/domain","i:/domain_preferences"); >>> - loads all AVPs with ID from 'domain_preferences' >>> table for domain from RURI >>> Q2: Is "$ruri" the caller user's URI? and the "domain" is the same? >> >> Yes. >> >>> Is "domain_preferences" table name? >> Yes, you can opt to select a different table. It is not necessary >> though. >> >>> Is it equal to my >>> "usr_preferences"? Or it is just a virtual table in memory? >> No and no. >> >> avpops is a very powerful module with various ways of using tables, >> loading av pairs and doing tests. Remember that avpairs are no more >> than pairs of attribute names (either string names as above or >> integer values i:34) and values (either string values or integer >> values). These avpairs must be loaded from somewhere, using >> avp_db_load avp_radius_load, etc using $from or $ruri for looking up >> the correct pair. >> >> g-)

19 years, 9 months

[Serusers] Re: Serusers Digest, Vol 22, Issue 23

by Brahim

hello i try to make run a web application clik to dial via SER , the server apache and SER is installed in to differents machines, but my programm bloc , i m ussing serfifo to communicate with ser. any help thanks.

19 years, 9 months

[Serusers] How to use avp_aliases?

by Charles Wang

Dear ALL: I have defined these lines at my ser.cfg. modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") 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") Q1: If I use usr_preferences as default table, should I insert some initial data to it? Can you give me some example? Schema List below: mysql> desc usr_preferences; +-----------+---------------+------+-----+---------+-------+ | Field | Type | Null | Key | Default | Extra | +-----------+---------------+------+-----+---------+-------+ | uuid | varchar(64) | | | | | | username | varchar(100) | | PRI | | | | domain | varchar(128) | | PRI | | | | attribute | varchar(32) | | PRI | | | | value | varchar(128) | | PRI | | | | type | int(11) | | | 0 | | | modified | timestamp(14) | YES | | NULL | | +-----------+---------------+------+-----+---------+-------+ And I have refered to http://www.voice-system.ro/docs/avpops And find many examples in it. But I still can't understand what it mean about avp_db_load(). my avp table is named as "usr_preferences" and it is empty before I use avpops modules. For this web page said: avp_db_load("$ruri/domain","i:/domain_preferences"); - loads all AVPs with ID from 'domain_preferences' table for domain from RURI Q2: Is "$ruri" the caller user's URI? and the "domain" is the same? Is "domain_preferences" table name? Is it equal to my "usr_preferences"? Or it is just a virtual table in memory? Thank you for your answer my stupid questions. Best Regard Charles

19 years, 9 months

[Serusers] status codes

by Mik Cheez

Is there a way to log status codes in SER? I would like to report the following types of info: SIP/2.0 404 Not Found SIP/2.0 200 OK In addition perhaps the callerid, uri, etc. :M

19 years, 9 months

[Serusers] MediaProxy Error Message "can't connect to MediaProxy"

by Charles Wang

Dear ALL: I build a ser and use a 0.9.0 example of ser.cfg. Now I have some error message in my log during I make a call. Feb 19 05:53:51 ser /usr/local/sbin/ser[3551]: SER: Load ACL Feb 19 05:53:51 ser /usr/local/sbin/ser[3551]: SER: a INT user Feb 19 05:53:51 ser /usr/local/sbin/ser[3549]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Feb 19 05:53:51 ser /usr/local/sbin/ser[3551]: SER: a FREE-PSTN user Feb 19 05:53:51 ser /usr/local/sbin/ser[3551]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Feb 19 05:53:57 ser /usr/local/sbin/ser[3549]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Feb 19 05:54:00 ser /usr/local/sbin/ser[3551]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Feb 19 05:54:11 ser /usr/local/sbin/ser[3550]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Thank you for your kind. Best Regard Charles My ser.cfg list below: # # $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=61.220.190.248 alias=ser.xxx.net alias=ser alias=xx.xx.xx.xx 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" # ------------------ 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! 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 -- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line # ------------- usrloc parameters # 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 -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # 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 parameters 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") # -- pft 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 parameters 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") # ------------------------- 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; }; # ------------------------------------------------------------------------ # 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); }; # ------------------------------------------------------------------------ # Registration Section # ------------------------------------------------------------------------ if (method=="REGISTER") { 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; }; # ------------------------------------------------------------------------ # NAT Test Section #1 # ------------------------------------------------------------------------ if (client_nat_test("3") && !search("^Record-Route:")) { force_rport(); fix_contact(); }; # ------------------------------------------------------------------------ # PSTN Section # ------------------------------------------------------------------------ if (method=="INVITE") { ## Deny PSTN to 0204 and 095 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); # local domain call break; }; # ------------------------------------------------------------------------ # Load ACL Section # ------------------------------------------------------------------------ if (method=="INVITE" && !isflagset(14)) { log(1, "SER: Load ACL\n"); if (is_from_local() || is_uri_host_local()) { 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("$ruri/username", "s:anoncallrej")) { if (avp_check("s:anoncallrej", "eq/y/i")) { log(1, "SER: a ANON-CALL-REJ user\n"); setflag(24); }; }; }; # ------------------------------------------------------------------------ # Anonymous Call Rejection Section # 24 # ------------------------------------------------------------------------ if (isflagset(24) && (method=="INVITE") && search("^(f|F)rom:.*(a|A)nonymous")) { route(8); break; }; # ------------------------------------------------------------------------ # Speed Dialing Section # ------------------------------------------------------------------------ if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) { sd_lookup("speed_dial"); }; # ------------------------------------------------------------------------ # 002 International Call Section # 29 # ------------------------------------------------------------------------ if (method=="INVITE" && uri=~"^sip:002[0-9]*@") { if (isflagset(29)) { route(6); } else { sl_send_reply("503", "Service Unavailable"); }; break; }; # ------------------------------------------------------------------------ # 0XXXXXXXXX Domestic Call Section # 29 # ------------------------------------------------------------------------ if (method=="INVITE" && uri=~"^sip:0[0-9]{9}@") { if (isflagset(29)) { route(6); } else { sl_send_reply("503", "Service Unavailable"); }; break; }; # ------------------------------------------------------------------------ # URI Compare Section # # Here we compare the "from" and "to" to see if the caller is dialing # their own extension. If so then we route to voicemail(31) if needed # ------------------------------------------------------------------------ if (method=="INVITE") { avp_write("$from", "i:34"); if (avp_check("i:34", "eq/$ruri/i")) { if (isflagset(31)) { route(5); break; } else { sl_send_reply("486", "Busy"); break; }; }; }; # ------------------------------------------------------------------------ # Do Not Disturb Section # ------------------------------------------------------------------------ if (avp_db_load("$ruri/username", "s:donotdisturb")) { if (avp_check("s:donotdisturb", "eq/y/i")) { route(5); break; }; }; # ------------------------------------------------------------------------ # Blind Call Forwarding Section # ------------------------------------------------------------------------ if (method=="INVITE") { # 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 route(1); if (avp_check("$calltype", "eq/-/i")) { sl_send_reply("503", "Service Unavailable"); break; }; # test for domestic PSTN gateway if (avp_check("$calltype", "eq/dom/i")) { route(3); break; }; # test for international PSTN gateway if (avp_check("$calltype", "eq/int/i")) { route(6); break; }; }; }; # ------------------------------------------------------------------------ # Call Routing Section # ------------------------------------------------------------------------ if (!lookup("location")) { # if flag 31 (ie voicemail) is set and we made it here this means # the user's phone is not registered anywhere. We'll forward to # voicemail after this block because we need to check the call # forward settings first if (isflagset(31)) { # flag 19 means the user has voicemail but is not online # so we need to remember to send to voicemail if call # forwarding is not enabled setflag(19); }; if (method=="INVITE") { if (does_uri_exist()) { # subscriber record found, but they're offline sl_send_reply("480", "Temporarily Unavailable"); break; }; if (uri=~"^sip:0[0-9]{9}@") { # Send to PSTN Gateway if (isflagset(28)) { route(3); } else { sl_send_reply("503", "Service Unavailable"); }; break; }; sl_send_reply("404", "User Not Found"); break; }; }; # ------------------------------------------------------------------------ # Call Forwarding Section # ------------------------------------------------------------------------ if (method=="INVITE") { # save R-URI in a temp AVP for later use avp_write("$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 $calltype as follows: # # "-" = R-URI is not allowed # "dom" = R-RURI is a domestic call # "int" = R-RURI is an international call # "sip" = R-RURI is a sip call avp_write("-", "$calltype"); # Call rejected if ((uri=~"^sip:0204[0-9]*@") || (uri=~"^sip:095[0-9]*@") || (uri=~"^sip:[0-9]{3}@")) { break; }; lookup("aliases"); if (!lookup("location")) { if (uri=~"^sip:0[0-9]{9}@") { # test for domestic PSTN number if (isflagset(28)) { avp_write("dom", "$calltype"); }; # } else if (uri=~"^sip:011[0-9]*@") { } else if (uri=~"^sip:002[0-9]*@") { # test for international PSTN number if (isflagset(29)) { avp_write("int", "$calltype"); }; }; break; }; avp_write("sip", "$calltype"); } route[2] { 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 (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] { # All Domestic Calls Go To PSTN 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; }; }; # SIP->PSTN calls get 45 seconds to timeout avp_write("i:45", "inv_timeout"); rewritehost("xx.xx.xx.xx"); 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)) { t_on_failure("1"); }; t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[4] { # 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] { # 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("99.99.99.100:5060"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[6] { # All International Calls Go To PSTN 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; }; }; # SIP->PSTN calls get 45 seconds to timeout avp_write("i:45", "inv_timeout"); rewritehost("xx.xx.xx.xx"); 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] { # 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] { # 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] { # 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")) { avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway route(3); } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway route(6); } else { # default to sip call 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_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway route(3); } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway route(6); } else { # default to sip call route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); route(4); break; }; }

19 years, 9 months

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sr-users February 2005