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 # ------------------ 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); }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); };
if (!uri==myself) { append_hf("P-hint: outbound\r\n"); 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 (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] { 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] { 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); }
El Thursday 09 August 2007 11:59:08 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.
Can't understand, your file has:
route[3] { if (uri=~"^sip:[0-9]{10}@") { xlog(" ---------------- Call to PSTN ---------------\n"); # <- DEBUG route(4); exit; }; ... } route[4] { rewritehost("172.20.14.32"); route(1); }
So the URI should be modified and you should see the xlog I've added above.
But of course note that the only is modified is the URI, and not the "To:" field that doesn't change (and doesn't must it).
Anybody can give me some advices?
First try adding that XLOG and test if it appears when calling to that number.
Regards.ç
Hi,
I think there is an error in the config. You always execute route[1] first and the call is executed. You should make in the invite query a differentiation between the numers that come in:
if (uri=~"^sip:[0-9]{10}@") { xlog(" ---------------- Call to PSTN ---------------\n"); # <- DEBUG route(4); exit; }; else { if (!t_relay()) { sl_reply_error(); }; exit; }
-----Original Message----- From: users-bounces@openser.org [mailto:users-bounces@openser.org] On Behalf Of Iñaki Baz Castillo Sent: Thursday, August 09, 2007 12:18 PM To: users@openser.org Subject: Re: [OpenSER-Users] Why if (uri==myself) is always true? (aboutopenser call a number PSTN)
El Thursday 09 August 2007 11:59:08 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.
Can't understand, your file has:
route[3] { if (uri=~"^sip:[0-9]{10}@") { xlog(" ---------------- Call to PSTN ---------------\n"); # <- DEBUG route(4); exit; }; ... } route[4] { rewritehost("172.20.14.32"); route(1); }
So the URI should be modified and you should see the xlog I've added above.
But of course note that the only is modified is the URI, and not the "To:" field that doesn't change (and doesn't must it).
Anybody can give me some advices?
First try adding that XLOG and test if it appears when calling to that number.
Regards.ç
-- Iñaki Baz Castillo ibc@in.ilimit.es
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