sr-users December 2004

sr-users@lists.kamailio.org

184 participants
400 discussions

by Felipe Martins

Hi everybody, It seems that i'm having a problem with my user authentication. I did the following. Added two users at ser # serctl add 1234 1234 1234(a)192.168.4.10 # serctl add 4321 1234 4321(a)192.168.4.10 I've checked my ser databases at subscriber table and the users were there the way I did. I've configured my ser.cfg with mysql.so (which I use for my users database), so as the auth modules auth.so and auth_db.so. MySQL is running, without errors, i've already created ser tables with ser_mysql.sh, the database and the tables were created without further problems. When I try to run ser with the default ser.cfg, my users get authenticated but when I use my ser.cfg configured with auth modules and mysql.so, they don't. When my equipament is trying to auth at ser, it gives me first a "401 Unauthorized" because the fist clear text password sent, and then it gives me "500 Server Internal Error" the Erros are attached below ------------ Errors ------------------- Sending SIP PDU to ( 192.168.4.10:5060 ) from 5060 REGISTER sip:192.168.4.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.30:5060;branch=z9hG4bK9307c100a43 From: <sip:1234@192.168.4.10>;tag=9307c100a4 To: sip:1234@192.168.4.10 Call-ID: 93526c07-ec2e-c1aa-8000-0002a40052bc(a)192.168.14.30 CSeq: 3 REGISTER Date: Thu, 01 Jan 1970 00:00:00 GMT User-Agent: AddPac SIP Gateway Contact: sip:1234@192.168.14.30 Expires: 3600 Content-Length: 0 Max-Forwards: 70 AP160(config-sip-ua)# Received SIP PDU from ( 192.168.4.10:5060 ) SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.14.30:5060;branch=z9hG4bK9307c100a43 From: <sip:1234@192.168.4.10>;tag=9307c100a4 To: sip:1234@192.168.4.10;tag=f967824f7e9189ff49a9fc75e4ea4f1c.cb2a Call-ID: 93526c07-ec2e-c1aa-8000-0002a40052bc(a)192.168.14.30 CSeq: 3 REGISTER WWW-Authenticate: Digest realm="192.168.4.10", nonce="3ee75c005bedc1108e8c7db3c737 4b74140e97c4" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 192.168.4.10:5060 "Noisy feedback tells: pid=3788 req_src_ip=192.168 .14.30 req_src_port=5060 in_uri=sip:192.168.4.10 out_uri=sip:192.168.4.10 via_cnt==1" Sending SIP PDU to ( 192.168.4.10:5060 ) from 5060 REGISTER sip:192.168.4.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.30:5060;branch=z9hG4bK9307c100a44 From: <sip:1234@192.168.4.10>;tag=9307c100a4 To: sip:1234@192.168.4.10 Call-ID: 93526c07-ec2e-c1aa-8000-0002a40052bc(a)192.168.14.30 CSeq: 4 REGISTER Date: Thu, 01 Jan 1970 00:00:00 GMT User-Agent: AddPac SIP Gateway Authorization: Digest username="1234", realm="192.168.4.10", nonce="3ee75c005bedc1 108e8c7db3c7374b74140e97c4", uri="sip:192.168.4.10", response="33ef4638b299ca839d5 d03240822aeca", algorithm=MD5 Contact: sip:1234@192.168.14.30 Expires: 3600 Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 192.168.4.10:5060 ) SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 192.168.14.30:5060;branch=z9hG4bK9307c100a44 From: <sip:1234@192.168.4.10>;tag=9307c100a4 To: sip:1234@192.168.4.10;tag=f967824f7e9189ff49a9fc75e4ea4f1c.306d Call-ID: 93526c07-ec2e-c1aa-8000-0002a40052bc(a)192.168.14.30 CSeq: 4 REGISTER Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 192.168.4.10:5060 "Noisy feedback tells: pid=3789 req_src_ip=192.168 .14.30 req_src_port=5060 in_uri=sip:192.168.4.10 out_uri=sip:192.168.4.10 via_cnt==1" ------------------------------------------------------------------ At the moment I'm just testing the authentication features, before the routing ones, so my main problem is with it. What am I doing wrong ? I've attached my current ser.cfg configuration for more information. Thanks in advance. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

19 years, 6 months

[Serusers] Newbie User Auth Problem

by Felipe Martins

19 years, 6 months

[Serusers] auth in provider server

by tito

-----BEGIN PGP SIGNED MESSAGE----- Hash: MD5 Hello! I have installed with mysql support and works fine. Now, I want to make the user auth on it to redirect to the provider sip server. They need to auth on it but I want ser to get the auth too to know in every moment which users are connected and the status of their calls. I have been reading the docs and don't find how to do that. Could you give me some help? Thanks. -----BEGIN PGP SIGNATURE----- Version: 2.6 iQEVAwUAQcf9/3HoJ4bX5QlXAQHPzAgAwthefyFR/fbYE+wrrQaaHaD7ZQ0QnuKJ 3bBQ0z6/HZMNAkvM7vs6f2KG9nZ4Z+DkgZoG31Ob8DgS8a6hFQUvWgCPQV0pCIAn 3ivvTwK6wcC+vialroBkTDb2BwJVWgOqT7ZCK1AYucoKQLo0wLPtceKSE2S6cUkr xr0PjbiMrycV5myb1BJmmDJmdV/UIeVxqqHMBd50zNn8TLmAUKyd4qdg/ZA4pjTY h3iFR9lEdcq30bb+Ozu5nZ6U5x3dL8ADjk/NxstOalS9yB+0KDbY4b/K7VG5TQg7 vN7Jh1B2AGj1w4xsOR/8sEKu6cQ+CHKm77a5RaNMMW59/eEKiOOhXg== =NoPb -----END PGP SIGNATURE-----

19 years, 6 months

[Serusers] Newbie problem

by O-Zone

Hi all, i'm trying to use SER with Radius auth for users. First i've tried to use IT with MySQL Auth and all works well so i decide to recompile all with Radius support and try in that way. I'm using FreeRadius with radiusclient and, after some tries, i'm able to compile SER with Radius support (following SER Radius Howto). Then i modify ser.conf in that way: root@cerberus:/usr/local/etc/ser# cat ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ 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" # 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" # RADIUS loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" #loadmodule "/usr/local/lib/ser/modules/uri_radius.so" # ----------------- setting module-specific parameters --------------- # modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- 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) modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } root@cerberus:/usr/local/etc/ser# but SER don't want to start ! When i do ser -c i have: root@cerberus:/usr/local/etc/ser# ser -c 0(8766) WARNING: could not rev. resolve 192.168.1.203 Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.1.203 [192.168.1.203]:5060 Aliases: localhost:5060 config file ok, exiting... root@cerberus:/usr/local/etc/ser# N.B.: (i'm trying to use it in a closed environment) but when i launch it: root@cerberus:/usr/local/etc/ser# ser Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.1.203 [192.168.1.203]:5060 Aliases: localhost:5060 root@cerberus:/usr/local/etc/ser# i have, in /var/log/syslog, that message: Dec 21 12:30:07 cerberus ser: WARNING: could not rev. resolve 192.168.1.203 and nothing else. SER is not running: root@cerberus:/usr/local/etc/ser# ps -ax | grep ser - 8788 pts/1 S+ 0:00 grep ser - root@cerberus:/usr/local/etc/ser# Someone can help me ? Thanks ! Oz -- ------ O-Zone ! www.zerozone.it

19 years, 6 months

[Serusers] www_challenge and transaction handling

by Klaus Darilion

Hi! I've some troubles with the radius authentication. If the radius authentication takes longer than 0.5 secondes, the client retransmits the message and causes another radius request. I tried to catch the retransmissions using t_newtran (ser.cfg snippet at the end of this email). This works fine for messages with credentials, but for the initial REGISTER messages (without credentials), which will be answered by www_challenge(), this causes the following warning: WARNING: script writer didn't release transaction Looks like www_challenge works only stateless. Is there a way to handle the challange stateful? Btw: I'm still using 0.8.12 - is this solved in newer versions? regards, klaus if ( !t_newtran()) { sl_reply_error(); xlog("L_ERR", "error creating new transaction\n"); break; }; xlog("L_INFO", "creating new transaction ... done\n"); if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; if (!check_to()) { if (!t_reply("403", "Forbidden - please use proper To")) { sl_reply_error(); }; break; }; if(!save("location")) { if (!t_reply("500", "Error saving contact")) { sl_reply_error(); }; break; }; break;

19 years, 6 months

[Serusers] smpp support

by Rao

Does ser have support for SMPP (Small Message Peer to Peer) ? What protocl does sms module currently uses to talk to the sms gateway. Thanks. Rao __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 6 months

[Serusers] Need explaination on SER with Asterisk

by Nikita Kang

Hi all, I'm very new to SER. Currently, I have an asterisk server running ,and am able make calls thru a conventional phone via an IAD box. Presently, I am trying to implement SER on top of asterisk. My plan is to use SER as a SIP Registrar and then route calls to asterisk. It needs to be stateful. Need some clarification on this: 1. When a call is made, (ngrep)it send an INVITE request instead of a REGISTER. why? 2. I then checked and found out that my sip user has already been stored in the "location" table even before any calls were made. Why? 3.What's the difference between "location" and "subscriber" table? Why does the if(!lookup("location")) always returns true and gives a NOT FOUND error when i can see my user in location table? Below is my ser.cfg file: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; # if the request is for other domain use # UsrLoc (in case, it does not work, use the # following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("202.171.60.3", "subscriber")) { www_challenge("202.171.60.3", "0"); break; } save("location"); break; } else { # NO registered contacts,exit now if (!lookup("location")) { sl_send_reply("404","Not Found"); break; }; } t_relay(); }; # forward to current uri now; use stateful # forwarding; that works reliably even if we # forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } I'm sure there's something wrong with my route logic. Please help. Nikki K __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 6 months

[Serusers] change port -> registration fail

by Thomas

[please ignore my previous message] Hi everyone, When I try to change the port from 5060 to other value (e.g. 1234), after restarting SER, the gateway (sipura 2000) fails to register even I change the "SIP port" to 1234. I use serctl ps to view the server process, only port 1234 is used for listening. I use ethereal to trace the packets, when the client sends the REQUEST, my SER server did not reply. Only 5060 works successfully. Do you have any ideas? Thomas

19 years, 6 months

[Serusers] change port -> registration fail!

by Thomas

Hi, When I try to change the port from 5060 to other value (e.g. 1234), after restarting SER, the gateway (sipura 2000) fails to register even I change the "SIP port" to 1234. I use serctl ps to view the server process, only port 80 is used for listening. I use ethereal to trace the packets, when the client sends the REQUEST, my SER server did not reply. Only 5060 works successfully. Do you have any ideas? Thomas On Fri, 2004-12-17 at 16:50, Atle Samuelsen wrote: > yup > > - atle > > * Thomas <support(a)cybertel.biz> [041217 09:27]: > > Hi, > > > > If I want to change the port 5060 to other port number, should I just > > modify ser.cfg and restart it? > > > > > > Thomas

19 years, 6 months

[Serusers] audio problem if forward to PSTN

by Marian Danisek

hello i have ser 0.8.14 working, some clients are behind nat others not. i have setup pstn gateway - asterisk a try to route some call here, but there is some problem with audio, from both clients - with real ip address and clients behind nat. Called party hear everything what caller say, but caller hear nothing. calling between client with real address and other behind nat works fine. ser and pstn-gateway have real ip addresses, they are no same subnet. Can anybody help me to solve this problem ? below is my sr.cfg best regards Marian # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ alias=mdk10.sunteq.sk alias=sunteq.sk #alias=atlas.sunteq.sk check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/lib/ser/modules/mysql.so" loadmodule "/lib/ser/modules/sl.so" loadmodule "/lib/ser/modules/tm.so" loadmodule "/lib/ser/modules/rr.so" loadmodule "/lib/ser/modules/maxfwd.so" loadmodule "/lib/ser/modules/usrloc.so" loadmodule "/lib/ser/modules/registrar.so" loadmodule "/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/lib/ser/modules/auth.so" loadmodule "/lib/ser/modules/auth_db.so" # load the voicemail module #loadmodule "/lib/ser/modules/vm.so" # load the enum module loadmodule "/lib/ser/modules/enum.so" # load the group module, to verify if a user forwards to voicemail loadmodule "/lib/ser/modules/group.so" # load the nathelper module loadmodule "/lib/ser/modules/nathelper.so" loadmodule "/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- registrar parameter # special NAT flag indicates that a registered client is behind NAT modparam("registrar", "nat_flag", 6) # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:heslo@localhost/ser") # -- 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") #modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- voicemail params -- #modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") # -- voicemail params -- #modparam("group", "db_url","mysql://serro:heslo@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 3) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #modparam("tm", "fr_inv_timer", 8 ) # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (nat_uac_test("2")) { log(1, "src address different than via header->NAT detected\n"); log(1, "force_rport and fix_nated_contact and setflag(5)\n"); #try NAT traversal, works only if the client is symmetrical force_rport(); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); # flag 5 indicates that incoming request is from NATed client setflag(5); }; if (method=="REGISTER") log(1, "REGISTER message received\n"); if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS message received\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len) { #if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; }; # Check for PSTN access if (uri=~"^sip:0[0-9]*@.*") { log(1, "going to PSTN route3\n"); route(3); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { log(1, "analyzing REGISTER request\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("mdk10.sunteq.sk", "subscriber")) { www_challenge("mdk10.sunteq.sk", "0"); break; }; if (isflagset(5)) { #register from nated client, save nat_flag=6 #in location table setflag(6); }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); #mark transaction for voicemail #if (is_user_in("Request-URI", "voicemail\n")) { # log(1, "requested user is in voicemail group"); # setflag(4); #}; # Process Aliases lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found log(1, "requested user not found\n"); route(4); break; }; }; #add failure route which should be performed if response code >=300 if (method=="INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1]{ log(1, "-------------------------------------------\n"); log(1, "entering route[1] - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7)"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; log(1, "relaying message ...\n"); if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { log(1, "-------------------------------------------\n"); log(1, "onreply_route[1] entered\n"); if (isflagset(6)) { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"100") ) { log(1, "status 100 received\n"); }; if ( (status=~"180") ) { log(1, "status 180 received\n"); }; if ( (status=~"202") ) { log(1, "status 202 received\n"); }; if ( (status=~"200" || status=~"183") ) { log(1, "status 2xx or 183"); if ( isflagset(7) ) { log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n"); force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } route[3] { if (method=="INVITE" && (!src_ip==194.1.222.26)) { log(1, "method is invite\n"); if (!proxy_authorize( "mdk10.sunteq.sk","subscriber")) { proxy_challenge( "mdk10.sunteq.sk", "0"); break; }; # let's check from=id ... avoids accounting confusion if(!is_user_in("credentials", "local")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN log(1, "authorized to PSTN\n"); # if you have passed through all the checks, let your call go to GW! force_rtp_proxy(); record_route(); t_on_reply("1"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("194.1.222.26","5060"); } route[4]{ log(1, "-------------------------------------------\n"); log(1, "entering route[4] = requested user not online\n"); # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == "BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n"); sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber and no echo/conference call if ( isflagset(4)) { log(1, "flag(4) active\n"); }; if (uri =~ "conference") { log(1, "conference call\n"); }; if (uri =~ "echo") { log(1, "echo call\n"); }; if ( !( isflagset(4) || (uri =~ "conference") || (uri =~ "echo") ) ) { log(1, "no voicemail subscriber->return 404"); sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; if ( isflagset(5) ) { log(1, "caller is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy"); force_rtp_proxy(); }; }; # forward to voicemail now rewritehostport("192.168.1.253:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("192.168.1.253", "5060"); } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); rewritehostport("212.17.35.184:5060"); append_branch(); t_relay_to_udp("212.17.35.184", "5060"); }

19 years, 6 months

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sr-users December 2004