Hello guys, does anyone have a VEGA gateway, SER and Grandstream phones setup that works? I am having some problems with outgoing calls. My call is being dropped after 16 seconds, while the incoming call is working properly. I have the latest firmware on my phones and here is my cnf file. 198.144.xxx.xxx - ser server 198.144.YYY.YYY - VEGA gateway 198.144.ZZZ.ZZZ - voicemail ( ser & sems ) # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/textops.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/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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") # time to give up on ringing -- global timer, applies to # all transactions modparam("tm", "fr_inv_timer", 20) # ------------------------- request routing logic ------------------- # main routing logic route { if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483", "Alas Too Many Hops"); break; }; if (!(method=="REGISTER")) record_route(); if (loose_route()) { t_relay(); break; }; if (!uri==myself) { t_relay(); break; } else { if (method == "REGISTER") { #if (!save("location")) { # sl_reply_error(); #}; # Uncomment this if you want to use digest authentication if (!www_authorize("198.144.xxx.xxx", "subscriber")) { www_challenge("198.144.xxx.xxx", "0"); break; }; save("location"); break; }; }; # Destination PSTN or H323? if( uri=~"^sip:9[0-9]*@198.144.xxx.xxx" ) { route(1); break; }; if( uri=~"^sip:\*74@198.144.xxx.xxx" ) { route(2); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc #if (is_user_in("Request-URI", "voicemail")) { # setflag(4); #}; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # if user is on-line and is in voicemail group, enable redirection if (method == "INVITE" ) { t_on_failure("1"); }; t_relay(); } # ------------ Send it to our PSTN ---------------------- route[1] { # Route to PSTN Gateways(s) if (uri=~"^sip:9[0-9]*@198.144.xxx.xxx") { ## This assumes that th e caller is log("Forwarding to PSTN\n"); ## registered in our re alm strip(1); t_relay_to_udp( "198.144.YYY.YYY", "5060" ); break; }; } route[2] { if (uri=~"^sip:\*74@198.144.xxx.xxx") { ## This assumes that the c aller is log("Picking up a Call on PSTN\n"); ## registered in our realm t_relay_to_udp( "198.144.YYY.YYY", "5060" ); break; }; } # ------------- handling of unavailable user ------------------ route[4] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber #if (!isflagset(4)) { # sl_send_reply("404", "Not Found and no voicemail turned on"); # break; #}; # forward to voicemail now rewritehostport("198.144.ZZZ.ZZZ:5090"); t_relay_to_udp("198.144. ZZZ.ZZZ", "5090"); #t_relay_to_tcp ("198.144. ZZZ.ZZZ","5090"); } # if forwarding downstream did not succeed, try voicemail running # at 198.144. ZZZ.ZZZ:5090 failure_route[1] { revert_uri(); rewritehostport("198.144. ZZZ.ZZZ:5090"); append_branch(); t_relay_to_udp("198.144. ZZZ.ZZZ", "5090"); #t_relay_to_tcp ("198.144. ZZZ.ZZZ","5090"); } Srbo Cvetkovic | CityNet, Inc. srbo(a)city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315