Hello,
I have a question. I have two users 1000 and 1001. I have two grandstream phones that I registered with those usernames. I have added aliases for them: serctl alias add 8000 sip:1000@mydomain.com serctl alias add 8000 sip:1001@mydomain.com
When i try to call 8000 from a softphone both of the phones ring. However after the timer is out I get 408: Request Timeout error instead of going to voicemail. I have a user 8000 with voicemail configured. my voicemail is running on a different machine though.
Any clues?
I tried adding it to the location table but still the same issue. I just want my phones to ring together, and if no answer go to voicemail. If it's picked up, then it will ring all phones again, so someone else can take the second call.I would like to use aliases so I we can still call local extensions internaly.
thanks in advance
Srbo Cvetkovic | CityNet, Inc. srbo@city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315
Srbislav Cvetkovic wrote:
Hello,
I have a question. I have two users 1000 and 1001. I have two grandstream phones that I registered with those usernames. I have added aliases for them: serctl alias add 8000 sip:1000@mydomain.com serctl alias add 8000 sip:1001@mydomain.com
When i try to call 8000 from a softphone both of the phones ring. However after the timer is out I get 408: Request Timeout error instead of going to voicemail. I have a user 8000 with voicemail configured. my voicemail is running on a different machine though.
voicemail for user 8000? 8000 ist just an alias - shouldn't be voicemail configured for user 1000 or 1001? Also your ser.cfg would be useful.
klaus
Any clues?
I tried adding it to the location table but still the same issue. I just want my phones to ring together, and if no answer go to voicemail. If it's picked up, then it will ring all phones again, so someone else can take the second call.I would like to use aliases so I we can still call local extensions internaly.
thanks in advance
Srbo Cvetkovic | CityNet, Inc. srbo@city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
OK here is my config file. Also you are right Klaus. There is no user 8000. It should just be an alias. And users 1001 and 1000 have voicemail configured. Once I take out the aliases everything works fine.
# # $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/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", 10)
# ------------------------- 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; };
if (method == "REGISTER") { if (!save("location")) { sl_reply_error(); }; break; };
# Destination PSTN or H323? if( uri=~"^sip:9[0-9]*@ mydomain.com" ) { route(1); break; };
if( uri=~"^sip:*74@mydomain.com" ) { 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"); # && isflagset(4) }; t_relay(); }
# ------------ Send it to our PSTN ---------------------- route[1] {
# Route to PSTN Gateways(s) if (uri=~"^sip:9[0-9]*@mydomain.com") { ## This assumes that th e caller is log("Forwarding to PSTN\n"); ## registered in our re alm strip(1); t_relay_to_udp( "gateway.com", "5060" ); break; };
}
route[2] {
if (uri=~"^sip:*74@mydomain.com") { ## This assumes that the c aller is log("Picking up a Call on PSTN\n"); ## registered in our realm t_relay_to_udp( "gateway.com", "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("voicemail.com:5090"); t_relay_to_udp("voicemail.com", "5090"); t_relay_to_tcp ("voicemail.com","5090"); }
# if forwarding downstream did not succeed, try voicemail running # at voicemail.com:5090
failure_route[1] { revert_uri(); rewritehostport("voicemail.com:5090"); append_branch(); t_relay_to_udp("voicemail.com", "5090"); t_relay_to_tcp ("voicemail.com","5090"); }
On Thursday, February 5, 2004, at 04:49 PM, Klaus Darilion wrote:
Srbislav Cvetkovic wrote:
Hello, I have a question. I have two users 1000 and 1001. I have two grandstream phones that I registered with those usernames. I have added aliases for them: serctl alias add 8000 sip:1000@mydomain.com serctl alias add 8000 sip:1001@mydomain.com When i try to call 8000 from a softphone both of the phones ring. However after the timer is out I get 408: Request Timeout error instead of going to voicemail. I have a user 8000 with voicemail configured. my voicemail is running on a different machine though.
voicemail for user 8000? 8000 ist just an alias - shouldn't be voicemail configured for user 1000 or 1001? Also your ser.cfg would be useful.
klaus
Any clues? I tried adding it to the location table but still the same issue. I just want my phones to ring together, and if no answer go to voicemail. If it's picked up, then it will ring all phones again, so someone else can take the second call.I would like to use aliases so I we can still call local extensions internaly. thanks in advance Srbo Cvetkovic | CityNet, Inc. srbo@city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315 _______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Srbo Cvetkovic | CityNet, Inc. srbo@city-net.com | Pittsburgh, PA voice: 412.481.5406 | fax: 412.431.1315