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(a)teigre.com>
To: "Tulika Pradhan" <tulikapradhan(a)hotmail.com>om>,
<serusers(a)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(a)lists.iptel.org
>
http://lists.iptel.org/mailman/listinfo/serusers
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