Hi,<br><br>I'm having some problems accounting missed serial forked calls to mysql database. <br><br>I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also SEMS voicemail which is forked as 3rd serial call leg if there is no answer at first two contacts.<br>
<br>For example, I have two users: <a href="mailto:oz@abc.hr" target="_blank">oz@abc.hr</a> and <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a>. <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a> also has set telephone number as alternative number if he is not reachable at <a href="mailto:sip%3Apero@abc.hr" target="_blank">sip:pero@abc.hr</a>. Moreover, <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a> has voicemail turned on. When <a href="mailto:oz@abc.hr" target="_blank">oz@abc.hr</a> calls <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a>, first <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a>'s SIP client rings, then if there is no answer and after the timeout telephone number rings and finally, if there is no answer at telephone and after the timeout INVITE is forked to SEMS.<br>
<br>There are two interesting scenarios accounting-wise which can happened:<br>1. <a href="mailto:oz@abc.hr" target="_blank">oz@abc.hr</a> calls <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a>, there are no answers and call is forked to voicemail.<br>
2. <a href="mailto:oz@abc.hr" target="_blank">oz@abc.hr</a> calls <a href="mailto:pero@abc.hr" target="_blank">pero@abc.hr</a>, there is no answer at SIP client, but pero answers call at telephone.<br>
<br>When scenario 1 happens, I want to have only one log (row) in missed_calls table.<br><br>When scenario 2 happens, I don't want to have a log in missed_calls table.<br><br>To accomplish this,<b> I want to log only the 2nd branch of the forked call. However, there is either a bug in acc module or I'm doing something wrong, and I can't get Kamailio to log only the 2nd branch</b>. I think that I am setting the FLT_ACCMISSED flag correctly - after the 2nd branch is handled and prior to calling the RELAY route. Logs show that FLT_ACCMISSED flag is set prior to calling t_relay(), and there are no errors in debug log. I am using $ru = "something" to rewrite URI prior to forking request. <br>
<br>I can easily set up logging of every call (two missed calls for serially
forked call to two locations) by setting FLT_ACCMISSED flag for each INVITE. I can set up logging of every call's 1st branch, by reseting FLT_ACCMISSED flag when handling 2nd branch of the call. Interestingly, logging of only the 2nd branch of the serial forked call works when there is no forking to voicemail! <br>
<br>Any ideas how to solve this problem? <br><br>Bellow are important parts of my config file. I'm running kamailio 3.1.4.<br><br>Cheers<br>Ozren<br><br><br># ----- acc params -----<br>/* what special events should be accounted ? */<br>
modparam("acc", "early_media", 0)<br>modparam("acc", "report_ack", 1)<br>modparam("acc", "report_cancels", 0)<br>modparam("acc", "detect_direction", 0)<br>
/* account triggers (flags) */<br>modparam("acc", "log_flag", FLT_ACC)<br>modparam("acc", "log_missed_flag", FLT_ACCMISSED)<br>modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)<br>
modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")<br>/* enhanced DB accounting */<br>#!ifdef WITH_ACCDB<br>modparam("acc", "db_flag", FLT_ACC)<br>
modparam("acc", "db_missed_flag", FLT_ACCMISSED)<br>modparam("acc", "db_url", DBURL)<br>modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$tU;dst_domain=$td;src_ip=$si")<br>
#!endif<br><br>...<br><br><br># Main SIP request routing logic<br># - processing of any incoming SIP request starts with this route<br>route {<br><br> # per request initial checks<br> route(REQINIT);<br><br>
if (src_ip != ****) {<br> # NAT detection<br> route(NAT);<br> }<br><br> # handle requests within SIP dialogs<br> route(WITHINDLG);<br><br> ### only initial requests (no To tag)<br>
<br> # CANCEL processing<br> if (is_method("CANCEL"))<br> {<br> if (t_check_trans())<br> t_relay();<br> exit;<br> }<br><br> t_check_trans();<br>
<br> # authentication<br> route(AUTH);<br><br> # record routing for dialog forming requests (in case they are routed)<br> # - remove preloaded route headers<br> remove_hf("Route");<br>
if (is_method("INVITE|SUBSCRIBE"))<br> record_route();<br><br> # account only INVITEs<br> if (is_method("INVITE"))<br> {<br> setflag(FLT_ACC); # do accounting<br>
}<br><br> # dispatch requests to foreign domains<br> route(SIPOUT);<br><br> ### requests for my local domains<br><br> # handle presence related requests<br> route(PRESENCE);<br><br>
# handle registrations<br> route(REGISTRAR);<br><br> if ($rU==$null)<br> {<br> # request with no Username in RURI<br> sl_send_reply("484","Address Incomplete");<br>
exit;<br> }<br><br> # dispatch destinations to PSTN<br> route(PSTN);<br><br> if ( is_method("INVITE") ) {<br> route(DBALIASES);<br> #check for user defined forking priorities and timers<br>
route(FORK);<br> }<br><br> # user location service<br> route(LOCATION);<br><br> route(RELAY);<br>}<br><br><br><br>#check for user defined forking priorities and timers<br>route[FORK]{<br>
sql_query("con", "select * from usr_pref_custom where uuid='$tu'", "pref");<br><br> $avp(uuid)=$dbr(pref=>[0,0]);<br> $avp(email)=$dbr(pref=>[0,1]);<br> $avp(prio1)=$dbr(pref=>[0,2]);<br>
$avp(prio2)=$dbr(pref=>[0,3]);<br> $avp(timer1)=$dbr(pref=>[0,5]);<br> $avp(timer2)=$dbr(pref=>[0,6]);<br><br> if (strlen($avp(prio1))>5) {<br><br> # user has multiple contacts, do serial forking<br>
setflag(FLT_USRPREF);<br><br> # set counter<br> if (!$avp(prio)) {<br> $avp(prio) = 1;<br> }<br><br> # overwrite request URI with highest priority contact<br>
if ($avp(prio1) =~ "^sip:00") {<br> $ru = $avp(prio1) + "@host";<br> xlog("L_INFO","PRIO 1 is tel number, RURI set: $ru");<br>
}<br> else {<br> $ru = $avp(prio1);<br> xlog("L_INFO","PRIO 1 is SIP URI, RURI set: $ru");<br> }<br> }<br>
}<br><br><br>route[RELAY] {<br>#!ifdef WITH_NAT<br> if (check_route_param("nat=yes")) {<br> setbflag(FLB_NATB);<br> }<br> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {<br>
route(RTPPROXY);<br> }<br>#!endif<br><br> /* example how to enable some additional event routes */<br> if (is_method("INVITE")) {<br><br> t_on_reply("REPLY_ONE");<br>
t_on_failure("FAIL_ONE");<br><br> #if users have priorities set, use FAIL_FORK failure route<br> if ( isflagset(FLT_USRPREF) ) {<br> t_on_failure("FAIL_FORK");<br>
}<br> }<br><br> if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS SET");<br> else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT SET");<br>
if (!t_relay()) {<br> sl_reply_error();<br> }<br> exit;<br>}<br><br><br># Handle requests within SIP dialogs<br>route[WITHINDLG] {<br> if (has_totag()) {<br> # sequential request withing a dialog should<br>
# take the path determined by record-routing<br> if (loose_route()) {<br> xlog("L_INFO","WITHINDLG, loose_route()");<br> if (is_method("BYE")) {<br>
xlog("L_INFO","WITHINDLG, BYE, DO ACCOUNTING");<br> setflag(FLT_ACC); # do accounting ...<br> setflag(FLT_ACCFAILED); # ... even if the transaction fails<br>
}<br> route(RELAY);<br> } else {<br> if (is_method("SUBSCRIBE") && uri == myself) {<br> # in-dialog subscribe requests<br>
route(PRESENCE);<br> exit;<br> }<br> if ( is_method("ACK") ) {<br> if ( t_check_trans() ) {<br>
# no loose-route, but stateful ACK;<br> # must be an ACK after a 487<br> # or e.g. 404 from upstream server<br>
t_relay();<br> exit;<br> } else {<br> # ACK without matching transaction ... ignore and discard<br>
exit;<br> }<br> }<br> sl_send_reply("404","Not here");<br> }<br> exit;<br>
}<br>}<br><br><br># USER location service<br>route[LOCATION] {<br><br> #skip if $ru is telephone number<br> if ($ru =~ "^sip:00") {<br> xlog("L_INFO","SKIP lookup...");<br>
}<br> else {<br> if (!lookup("location")) {<br> switch ($rc) {<br> case -1:<br> case -3:<br> t_newtran();<br>
t_reply("404", "Not Found");<br> exit;<br> case -2:<br> sl_send_reply("405", "Method Not Allowed");<br>
exit;<br> }<br> }<br> }<br><br> # when routing via usrloc, log the missed calls also, but only if user doesn't have prios set<br>
if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) {<br> setflag(FLT_ACCMISSED);<br> }<br>}<br><br><br># Failure route for forked calls<br>failure_route[FAIL_FORK] {<br>
#!ifdef WITH_NAT<br> if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) {<br> unforce_rtp_proxy();<br> }<br>#!endif<br><br> if ($avp(prio) >= 1) {<br>
$avp(prio) = $avp(prio) + 1;<br><br> # handle 2nd branch<br> if ( ($avp(prio) == 2) && ( isflagset(FLT_USRPREF) )) {<br> t_on_failure("FAIL_FORK");<br>
<br> if ($avp(prio2) =~ "^sip:00") {<br> xlog("L_INFO","FAIL FORK, PRIO 2 is tel number");<br> $ru = $avp(prio2) + "@host";<br>
}<br> else {<br> xlog("L_INFO","FAIL FORK, PRIO 2 is SIP URI");<br> $ru = $avp(prio2);<br> route(LOCATION);<br>
}<br> setflag(FLT_ACCMISSED);<br> }<br><br> else {<br> $avp(prio) = 0;<br> $ru = $(avp(uuid));<br>
rewritehostport("host:port");<br> xlog("L_INFO","FAIL FORK, VOICEMAIL email:$avp(email), ru:$ru, br: $br");<br> append_hf("P-App-Name: voicemail\r\n");<br>
append_hf("P-App-Param: Email-Address=$avp(email)\r\n");<br> }<br> route(RELAY);<br> }<br><br> if (t_is_canceled()) {<br> exit;<br>
}<br>}