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(a)gigo.co.uk>
To: Steve Blair <blairs(a)isc.upenn.edu>
CC: 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 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-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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Serusers mailing list
>>>>>serusers(a)lists.iptel.org
>>>>>http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
>>>>>.
>>>>>
>>>>
>>>>_______________________________________________
>>>>Serusers mailing list
>>>>serusers(a)lists.iptel.org
>>>>http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>>
>>>
>>>
>