[Serusers] SERWEB+SER help!!!!!

Here is my ser.cg file. I try to configure send IM, Missed calls, Voicemail. I just can configure Send IM. Now I can call isdngw, ivr,.. I need help to configure lookup location logic for missed calls and voicemail (offline users) Anybody could help me to configure the others functions (problem with logic !!). harry Regards # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.0.1 port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ 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/domain.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/textops.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_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- 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) # --registrar params-- modparam("registrar", "use_domain", 1) # --domain params-- modparam("domain", "db_url", "mysql://ser:heslo@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching # --acc params-- modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) # --msilo params-- modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@192.168.0.1") # --uri params-- #modparam("uri", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("uri", "subscriber_table", "subscriber") # --vm params-- modparam("voicemail", "db_url", "mysql://ser:heslo@localhost/ser") # ------------------------- 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"); break; }; if ( msg:len > 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; }; # --isdn gateway-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:[0-9]{10}@.*") { route(2); break; }; }; # --ivr conf-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:5000@.*") { route(4); break; }; }; # --conference-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:6000@.*") { route(5); break; }; }; # --play an annoucement-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:7000@.*") { route(6); 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 (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n"); # we have changed the R-URI with the contact address -- ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted for delivery"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } failure_route[4] { route(3); #append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } route[2] { # ############################## # # isdngw specific configuration # # ############################## # if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:[0-9]{10}@.*"){ if(!vm("/tmp/am_fifo","isdngw")){ log("could not contact isdngw\n"); t_reply("500","could not contact isdngw"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the isdngw. } else if(uri=~"sip:7000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[3] { ############################################ # Voicemail specific configuration - begin # ############################################ if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ if(!vm("/tmp/am_fifo","voicemail")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; break; }; if(method=="BYE" || method=="CANCEL") { if(!vm("/tmp/am_fifo","bye")) { log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; break; }; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[4] { ###################################### # ivr specific configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:5000@.*") { if (!vm("/tmp/am_fifo", "ivr")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact ivr server"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact ivr\n"); t_reply("500","could not contact ivr"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[5] { # #################################### # conference specific configuration # # #################################### if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:6000@.*"){ if(!vm("/tmp/am_fifo","conference")){ log("could not contact conference\n"); t_reply("500","could not contact conference"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the conference. } else if(uri=~"sip:5000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[6] { ###################################### # announcement configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:7000@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact announcement"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact annoucement\n"); t_reply("500","could not contact annoucement"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[7] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; if (t_newtran()) { t_reply("404", "Not Found"); acc_db_request("404 missed call", "missed_calls"); }; }