[Serusers] 408 to Caller UA when CANCEL to Callee

Tulika Pradhan tulikapradhan at hotmail.com
Fri Sep 16 12:41:37 CEST 2005


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 at 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 at 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 at teigre.com>
>To: "Tulika Pradhan" <tulikapradhan at hotmail.com>, <serusers at 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
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers at lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>





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