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:

what happens if you increase your timeout values, i.e send cancel before you get the timeout Iqbal Tulika Pradhan wrote: _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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" &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 >>>>> 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 > > > >

From: Iqbal &lt;iqbal(a)gigo.co.uk&gt; To: Steve Blair &lt;blairs(a)isc.upenn.edu&gt; CC: Tulika Pradhan &lt;tulikapradhan(a)hotmail.com&gt;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" &lt;greger(a)teigre.com&gt; >>>>>>To: "Tulika Pradhan" &lt;tulikapradhan(a)hotmail.com&gt;om>, &gt;&gt;&gt;&gt;&gt;&gt;&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;&gt;&gt;&gt;&gt;serusers(a)lists.iptel.org >>>>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>>_______________________________________________ >>>>>Serusers mailing list &gt;&gt;&gt;&gt;&gt;serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>>. >>>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list &gt;&gt;&gt;&gt;serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> >

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 >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers