Re: [Serusers] 408 to Caller UA when CANCEL to Callee

what happens if you increase your timeout values, i.e send cancel before you get the timeout

Iqbal

Tulika Pradhan wrote:

my ser.cfg file is attached below.

any help/pointers for what the problem may be would be great.

the problem comes when i dial anynumber starting with '3'

i want 8001211 to be dialed and if there is failure, then 8001210 to be dialed.

thanks,

tulika

# # $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) #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/acc.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" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters ---------------

# -- usrloc params --

modparam("usrloc", "db_mode", 2)

# -- auth params -- modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) # # 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("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("tm", "fr_inv_timer", 15) modparam("tm", "fr_timer", 10) # 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        # 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) record_route();
   if (uri==myself) {
    if (method=="REGISTER") {

           save("location");
           break;
    };

    if (method==INVITE) {
           if (uri=~"^sip:0[0-9]*@") {
                   log(1, "beginning with 0\n");
                   rewritehost("192.168.1.201");
                   rewriteport("5060");
                   t_relay_to_udp("192.168.1.201","5065");
                   break;
           } else if (uri=~"^sip:500@") {
                   log(1, "Accessing Voicemail\n");
                   setflag(1);
                   rewriteport("5065");                } else if 

(uri=~"^sip:3[0-9]*@203.197.212.208") { # call hunt numbers beginning with 3 log(1, "beginning with 3\n"); seturi("sip:8001211@192.168.1.201"); append_hf("P-hint: call hunt\r\n"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> <%tu>\n"); t_on_failure("1"); t_relay();

           }
           if (!lookup("location")) {
               if (search("(P-hint): call hunt")) {
                                   log(1, "Call Hunt number not in 

location- Hangup\n"); #exec_msg("echo $SIP_OUSER >> /root/temp; echo $SIP_USER >> /root/temp; echo $SIP_OURI >> /root/temp; echo $SIP_RURI >> /root/temp"); # goto next number

exec_dset("/etc/ser/getnextnumber1 $SIP_OUSER; echo>/dev/null;"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> <%tu>\n"); t_relay(); } else { log(1, "Asterisk forwarding as user not logged in..\n"); rewritehost("192.168.1.201"); rewriteport("5065"); t_relay_to_udp("192.168.1.201","5065"); break; }

           }
          t_on_failure("1");

    }
  }
  if (!t_relay()) {
           sl_reply_error();
  };

}

failure_route[1] { log(1,"Failure 1\n");

   if (search("(P-hint): call hunt")) {
           log(1, "Call Hunt number failure - Hangup\n");
           append_branch("sip:8001210@192.168.1.201");
           t_on_failure("2");
           xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 

<%tu>\n"); t_relay(); } else { log(1, "Asterisk forwarding ..\n"); revert_uri(); rewritehostport(192.168.1.201:5065"); append_branch(); t_relay(); } }

failure_route[2] { # log (1, "in failure route 2\n"); }

    }
  }
  if (!t_relay()) {

                   t_relay_to_udp("192.168.1.201","5065");
                   break;

   if (method!="REGISTER") record_route();

...

From: "Greger V. Teigre" greger@teigre.com To: "Tulika Pradhan" tulikapradhan@hotmail.com, serusers@lists.iptel.org Subject: Re: [Serusers] 408 to Caller UA when CANCEL to Callee Date: Fri, 16 Sep 2005 08:36:53 +0200

Tulika, This is not a function of SER, but your ser.cfg file. We have just released a new Getting Started document at onsip.org that you may use as a reference to identify why your ser.cfg causes a 408 to be sent. g-)

Tulika Pradhan wrote:

...

hi,

i am facing the following situation.

UA1 calls a user(UA2) who does not answer. the control comes to failure_route where i try another UA (UA3). but as UA3 rings, SER sends 408 Request timeout to UA1 and call gets disconnected.

this is the SIP message flow.

UA1 SER UA2 UA3 INVITE----------------> INVITE--------------> <----------------TRYING <----------------RINGING <------------------RINGING

                         CANCEL-------------->

<---------------------408

INVITE----------------------------------------> <---------------------487 ACK-------------------> <-----------------------OK

<-------------------------------------------TRYING

<--------------------------------------------RINGING

(but UA already has got the busy tone) and does not hear this ringing.

if 408 was not sent to UA1, then the call could have been established.

what is going wrong,

regards,

tulika

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