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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