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

From: Iqbal &lt;iqbal(a)gigo.co.uk&gt; To: Tulika Pradhan &lt;tulikapradhan(a)hotmail.com&gt; CC: greger(a)teigre.com, serusers(a)lists.iptel.org Subject: Re: [Serusers] 408 to Caller UA when CANCEL to Callee Date: Fri, 16 Sep 2005 12:13:45 +0100 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" &lt;greger(a)teigre.com&gt; >>To: "Tulika Pradhan" &lt;tulikapradhan(a)hotmail.com&gt;om>, &lt;serusers(a)lists.iptel.org&gt; >>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 &gt;&gt;&gt;serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >_______________________________________________ >Serusers mailing list &gt;serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >. >