Hola,
La razón por la que escribo esta vez es para ver si me pueden ayudar adirigir llamas hacia un voicemail configurado en Asterisk.
El escenario es el siguiente: hasta este punto tengo usuarios que se encuentran en LDAP, estos se autentican a través de un servidor Radius con OpenSER, una vez que tengo los usuarios en Openser estos pueden hacer llamadas SIP entre usuarios registrados asé como tambien realizar llamadas a la pstn utilizando para ello Asterisk como gateway, por otro lado los usuarios de OpenSER pueden consultar su buzón de voz al presionar *98, el buzón esta vacío porque no se como hacer que OpenSer redireccione la llamada a Asterisk en caso de que el cliente este ocupado o no disponible.
Se que tengo que usar el failure_route pero no se como aplicarlo,
Gracias en adelanto por la ayuda,
Mario F.
Mi openser.cfg es el siguiente:
debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode 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 listen=udp:192.168.1.11 alias="tesis.com" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_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 ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/openser/modules/mysql.so"
loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- 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)
###############PARAMETROS RADIUS # -- acc params --
modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("auth_radius", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf")
# -- group_radius params -- modparam("group_radius", "use_domain", 1)
# -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102")
# -- 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; };
# 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()) { append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); }
route(1); };
if(is_method("INVITE") && !has_totag())
{ # set the acc flags log(1,"-----> LLAMADA SIP <----- \n"); setflag(1); setflag(2); }
if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("tesis.com")) { www_challenge("tesis.com", "0"); exit; }; save("location"); exit; };
if(uri=~"sip:*98@.*") { #authorize if a call is going to PSTN xlog("L_INFO", "CALL: Call to check voicemail\n"); rewritehostport("192.168.1.10:5060"); };
if(uri=~"sip:041[2-6][0-9][0-9][0-9][0-9][0-9][0-9]+@") { #xlog("L_ERR", "LLAMANDO A PSTN\n"); # set gateway address "ASTERISK" log(1, "LLAMANDO A PSTN -----> Forwarding to Asterisk <----- \n"); rewritehostport("192.168.1.10:5060"); route(1); };
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")) { acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not 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; }
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Hola,
Se que tengo que usar el failure_route pero no se como aplicarlo,
Detecta el codigo de error que te interesa, por ejemplo 408, y dirige la llamada al servidor de voicemail reescribiendo el RURI.
sr-users-es@lists.kamailio.org