# # $Id: openser.cfg 1676 2007-02-21 13:16:34Z 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=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # listen="10.10.3.190" 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 "xlog.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" loadmodule "uri_db.so" loadmodule "auth.so" loadmodule "auth_radius.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- #tm timeout for voicemail params modparam("tm", "fr_timer", 25) modparam("tm", "fr_inv_timer", 25) modparam("tm", "noisy_ctimer", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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 -- modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # 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 #Routing Script route { #check for old messages: could mean a problem with the DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; }; #check for extremely large messages; we don't need a sip dos attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, >= 2048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; #do not send to voicemail if BYE or CANCEL #is used to end call before user pickup or timeout if(method=="CANCEL" || method=="BYE") { setflag(10); }; #grant route if route headers already present if (loose_route()) { route(1); return; }; #Always require authentication, which could result in a PSTN, ie $$$ if (method=="REGISTER") { if(!radius_www_authorize("caerj.proderj.rj.gov.br")) { www_challenge("caerj.proderj.rj.gov.br", "0"); return; } else { #Save into user database, used below when checking if user is available xlog("L_INFO", "REGISTER: User Authenticated Correctly\n"); save("location"); return; }; }; if (method=="INVITE") { if (uri=~"sip:011[0-9]+@.*") { #authorize if a call is going to PSTN # if(!proxy_authorize("caerj.proderj.rj.gov.br", "subscriber")) # { # proxy_challenge("caerj.proderj.rj.gov.br", "0"); # return; # }; xlog("L_INFO", "CALL: Call to international number\n"); rewritehostport("127.0.0.1:5070"); } else if(uri=~"sip:\*86@.*") { #authorize if a call is going to PSTN #if(!proxy_authorize("domain.net", "subscriber")) #{ # proxy_challenge("domain.net", "0"); # return; #}; xlog("L_INFO", "CALL: Call to check voicemail\n"); rewritehostport("127.0.0.1:5070"); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "ERROR: User Not Found\n"); return; }; } else { #authorize if a call is going to PSTN #if(!proxy_authorize("domain.net", "subscriber")) #{ # proxy_challenge("domain.net", "0"); # return; #}; #Call destination is PSTN, so send it to the gateway (Net.com) xlog("L_INFO", "CALL: PSTN gateway\n"); rewritehostport("127.0.0.1:5070"); }; }; #Make sure that all subsequent requests go through us; record_route(); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "ERROR: User Not Found\n"); return; }; } else { #Call destination is PSTN, so send it to the gateway (Net.com) xlog("L_INFO", "CALL: PSTN gateway\n"); rewritehostport("127.0.0.1:5070"); }; record_route(); }; #ALL PROCESSING IS DONE, SO ROUTE route(1); } route[1] { #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } failure_route[2] { if(!t_was_cancelled()) { revert_uri(); rewritehostport("127.0.0.1:5070"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } }