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@city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315