hello I have problems to make to operate the voicemail. when it is accomplished the call not lets me to record the message and is lost the ans_machine.
Español
hola tengo problemas para hacer funcionar el voicemail. cuando se realiza la llamada no me deja grabar el mensaje y se pierde el ans_machine.
estos son mis archivos de configuracion.
Log messages
Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ACC: call missed: from=Save1sip:102@mydomain.com;tag=28778-a725, i-uri=sip:100@mydomain.com, method=INVITE, o-uri=sip:100@262.61.161.129:5060;user=phone, code=488 Not Acceptable Here Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: failure_route[1]:jump to route[4]:vm Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): You have an error in your SQL syntax near ''' at line 1 Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): Error while submitting query Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: db_query() failed. Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: vm_get_user_info failed Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: route[3]:vm:voicemail failed Oct 28 17:14:09 chitara /usr/sbin/ser[21047]: route[]:eof Oct 28 17:14:10 chitara /usr/sbin/ser[21073]: route[]:eof ---------------------------------------------------------------------------- ------------------------------ SER.CFG # # $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $ # # config script with voicemail, PSTN dial-out functionality #
# ----------- global configuration parameters ------------------------
debug=1 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E)
# Uncomment these lines to enter debugging mode /* debug=8 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 fifo="/tmp/ser_fifo" sip_warning=no listen=262.69.20.92 alias=sip.boxip.com.ar # # ------------------ module loading ---------------------------------- # # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" # loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/textops.so" # loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.so" # # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/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) # # -- 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") # # modparam("tm", "fr_inv_timer", 30) #INVITE timeout modparam("tm", "fr_timer", 15) #negative INVITE reply or no #final reply for a request for ACK # modparam("voicemail", "db_url", "sql://ser:heslo@localhost/ser") # modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # modparam("acc", "log_level", 1) modparam("acc", "log_flag", 2) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_fmt", "fimos") #
# ------------------------- request routing logic ------------------- # # main routing logic # alias=262.69.20.92 #sip server IP address alias=mydomain.com #sip server FQDN # route{ # log(1,"entering main route"); setflag(2); #set flag for accounting
# 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"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; };
# 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; };
# 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") { # digest authentication # log(1,"request for registration"); # if (!www_authorize("mydomain.com", "subscriber")) { # www_challenge("mydomain.com", "0"); # break; # }; save("location"); break; };
/* ********** Dial out to PSTN logic ************* */
#forward 411[information] and 911[emergency] requests to gateway if(uri=~"^sip:(9|2)11@(boxip.com.ar|200.69.20.92)"){ log(1,"411/911 expression match"); route(2); break; }; #forward numerical 7 digit requests to gateway if(uri=~"^sip:[0-9]{7}@(mydomain.com|262.69.20.92)"){ log(1,"7 digit expression match"); route(2); break; }; # strip 650 and forward to GW if user dials 650 before phone no. if(uri=~"^sip:650[0-9]{7}@mydomain.com|262.69.20.92)"){ strip(3); log(1,"650 area code dialed, 650 stripped"); route(2); break; }; #forward numerical 10 digit requests to gateway, append a 1 first if(uri=~"^sip:[0-9]{10}@(mydomain.com|262.69.20.92)"){ prefix("1"); log(1,"10 digit expression match, prefix 1"); route(2); break; }; #forward numerical 11 digit requests that start with a 1 to GW if(uri=~"^sip:1[0-9]{10}@(mydomain.com|262.69.20.92)"){ log(1,"10 digit exp match w/leading 1"); route(2); break; }; #forward international N digit requests to gateway if(uri=~"^sip:011[0-9]+@(mydomain.com|262.69.20.92)"){ log(1,"international expression match"); route(2); break; };
/* ********** VOICEMAIL logic ************* */ if (uri=~"^sip:voicemail+@"){ log(1,"sip:voicemail uri match"); route(3); break; };
/* ****** Find Aliases and Locations of users ********* */ #lookup "aliases" before looking up "location" lookup("aliases");
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { route(3); break; # this section needs help. I need to differentiate between users that # exits buy aren't online (send these requests to voicemail) # requests for users that aren't in my subscriber database # (send these requests a 404 reply) sl_send_reply("404", "User Not Found"); break; }; };
# forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP
t_on_failure("1");
if (!t_relay()) { sl_reply_error(); }; log(1,"route[]:eof"); }
route[2]{ log(1,"route[2]:SIP-to-PSTN call routed"); rewritehostport("262.69.20.92:5060"); if(!t_relay()){ sl_reply_error(); }; }
route[3]{ log(1,"route[3]:vm:1"); if (method=="INVITE" || method=="BYE" || method=="REFER"){ log(1,"route[3]:vm:2");
if(t_newtran()){ t_reply("100","Trying -- just a second"); if(method=="INVITE" || method=="REFER"){ log(1,"route[3]:method==INVITE || REFER"); if(uri =~ "conference" ){
if(!vm("/tmp/am_fifo","conference")){
log(1,"route[3]:vm:conference failed"); t_reply("500","could not contact conference server"); }; } else if (uri =~"echo"){ if(!vm("/tmp/am_fifo","echo")){ log(1,"route[3]:vm:echo failed"); t_reply("500","could not contact echo"); }; } else{ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }; break; }; if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; break; }; } else{ log(1,"route[3]:vm:new transaction failed"); sl_send_reply("500", "new transaction failed"); break; }; }; }
route[4]{ # this should be voicemail logic that is specific to a failure_route # i.e. line busy, or timeout after a certain period with no answer if(method=="INVITE" || method=="REFER"){ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }else if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; };
}
failure_route[1]{
log(1,"failure_route[1]:jump to route[4]:vm"); # append_branch("sip:info at mydomain.com"); route(4); }
---------------------------------------------------------------------------- ------------------ SEMS.CFG
# optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/lib/sems/audio/
# optional parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav
# optional parameter: max_record=<seconds> # # - maximum record time max_record=30
# optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=262.61.161.129
# optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25
################################## # module specific parameters # ##################################
# add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end ---------------------------------------------------------------------------- ---------------------------
Thanks Gracias
Saludos
A su Disposición. Sergio Gerardo Festa