hello,
that was a problem when i did cut and paste from the ser.cfg, it should be as below,
} else if (uri=~"^sip:500@") { log(1, "Accessing Voicemail\n"); setflag(1); rewriteport("5065"); t_relay_to_udp("192.168.1.201","5065"); break;
} else if (uri=~"^sip:3[0-9]*@192.168.1.201") { # 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(); }
-tulika
From: Iqbal iqbal@gigo.co.uk To: Steve Blair blairs@isc.upenn.edu CC: 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 14:34:06 +0100
:-), Friday, heavy lunch
Iqbal
Steve Blair wrote:
Look at the middle of his routing statements. There is the following section of code :
---- cut here ---- } else if (uri=~"^sip:500@") { log(1, "Accessing Voicemail\n"); setflag(1); rewriteport("5065"); } else if(uri=~"^si"^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");
--- end cut ----See the line beginning with "rewriteport("5065")" ?
Iqbal wrote:
did I miss it...where ?
Steve Blair wrote:
You config file shows "(uri=~"^si"^sip:3[0-9]*@203.197.212.208") " is this really what you meant? I think the first "^si" is a typo is it not?
-Steve
Iqbal wrote:
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-routeprocessing 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 notin 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 >> >> >>_______________________________________________ >>Serusers mailing list >>serusers@lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > > > >
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