Hello everybody,
I want that openser can call a PSTN number for example: 0296487631. But in the request of INVITE, I see that the callee is always 0296487631@opensersip.test.machine ( and opensersip.test.machine is my sip server), that means the (uri==myself) is always ture. So this request of INVITE can not reache the gateway of PSTN. Anybody can give me some advices? This is my openser.cfg and I use openser+radius+mysql. Thank you very much!!!
Yin
# # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $
# simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. #
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4
# Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes #
port=5060
# uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem"
# ------------------ module loading ----------------------------------
#set module path mpath="/usr/local/lib/openser/modules/"
# Uncomment this if you want to use SQL database loadmodule "mysql.so"
loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" #loadmodule "auth_db.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "acc.so"
# ----------------- setting module-specific parameters ---------------
# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
# -- usrloc params --
modparam("usrloc", "db_mode", 0) modparam("auth_radius|group_radius|acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius","use_domain", 0) modparam("auth_radius", "service_type", 15)
modparam("auth", "secret", "testing123") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "User-Name=$fU;Sip-Src-IP=$si;Sip-Src-Port=$sp")
# 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)
# ------------------------- 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"); exit; };
if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; };
#-------set flag for Radius Accounting #if (!method=="OPTIONS") setflag(3);
#if (method=="INVITE"){ # log (1, "INVITE MESSAGE RECEIVED - START ACC\n"); # setflag(1); # setflag(2); #};
if (method=="BYE"){ log (1, "BYE - STOP ACCOUNTING\n"); setflag(1); };
if (method=="CANCEL"){ log (1, "CANCEL -STOP ACCOUNTING\n"); setflag(1); }; # 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();
# subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); };
if (!uri==myself) { # mark routing logic in request # append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} # route(1);
if (method=="ACK") { route(1); exit; }else if (method=="CANCEL"){ route(1); exit; }else if (method=="INVITE") { route(3); exit; }else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(1); exit; };
if (!lookup("location")) { sl_send_reply("404","Not User PSTN Found"); exit; }; route(1); }
# 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") {
# Uncomment this if you want to use digest authentication if (!radius_www_authorize("opensersip.test.machine")) { www_challenge("opensersip.test.machine", "0"); exit; };
save("location"); exit; };
if (method=="INVITE") {
if (!radius_proxy_authorize("opensersip.test.machine")){ proxy_challenge("opensersip.test.machine","0"); exit; }; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not SIP User Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); };
route(1); }
route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }
route[2] { # Register Message Handler sl_send_reply("100","Trying"); if (!radius_www_authorize("opensersip.test.machine")) { www_challenge("opensersip.test.machine", "0"); exit; }; if (!save("location")) { sl_reply_error(); }; }
route[3] { # Invite Message Handler # if (!radius_proxy_authorize("opensersip.test.machine")){ # proxy_challenge("opensersip.test.machine","0"); # exit; # }
# lookup("aliases"); # if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); # }; if (uri=~"^sip:[0-9]{10}@") { route(4); exit; };
if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); exit; }; sl_send_reply("404","User not Found"); exit; }; route(1); }
route[4] { rewritehost("172.20.14.32"); route(1); }
Hi Yin,
first try to use some xlog() messages to debug your script - see if it gets to your test, print the RURI to see it, etc. Note the uri==myself tests only the domain part (if the domain from uri is served by the proy). To see the domains served by proxy, see the aliases list dumped by openser at startup.
regards, bogdan
zze-WU Yin RD-MAPS-LAN wrote:
Hello everybody,
I want that openser can call a PSTN number for example:
0296487631. But in the request of INVITE, I see that the callee is always
0296487631@opensersip.test.machine ( and opensersip.test.machine is my sip server), that means the (uri==myself) is always ture. So this request of INVITE can not reache the gateway of PSTN. Anybody can give me some advices? This is my openser.cfg and I use openser+radius+mysql.
Thank you very much!!!
Yin
El Thursday 09 August 2007 11:54:12 zze-WU Yin RD-MAPS-LAN escribió:
Hello everybody,
I want that openser can call a PSTN number for example: 0296487631. But in the request of INVITE, I see that the callee is always 0296487631@opensersip.test.machine ( and opensersip.test.machine is my sip server), that means the (uri==myself) is always ture. So this request of INVITE can not reache the gateway of PSTN.
Not at all, you have a wrong concept. Of course the domain will be your OpenSer's domain. So after authenticate users and so you should "rewriteuri(SIP2PSTN_machine)", that's all.