[Serusers] 408 to Caller UA when CANCEL to Callee
Iqbal
iqbal at gigo.co.uk
Fri Sep 16 13:13:45 CEST 2005
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 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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>
> .
>
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