Hi Daniel, <br><br>thanks for the quick fix and reply.<br><br>What is the easiest way to try this new patch? I'm running kamailio 3.1.4 and there is no t_flush_flags() in tmx module in that version. I suppose I need to install Kamailio Devel from git (<a href="http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git">http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git</a>) to get t_flush flags() and your patch or is there a workaround to apply them to my 3.1.4 branch?<br>
<br>Cheers<br>Ozren<br><br><br><div class="gmail_quote">On Tue, Sep 6, 2011 at 2:18 PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com">miconda@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<br>
<br>
can you use t_flush_flags() after setting the accounting flag in
falure_route? Automatic update was missing so far, reported by Alex
Hermann as well. I just did a patch, so if you want to try it, see
the commit:<br>
<br>
<a href="http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c589ca35b2aa3097a3c9e2a5a050514337300c05" target="_blank">http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c589ca35b2aa3097a3c9e2a5a050514337300c05</a><br>
<br>
Actually, reporting if all goes fine with this patch, will help in
backporting it to 3.1 branch.<br>
<br>
Thanks,<br>
Daniel<div><div></div><div class="h5"><br>
<br>
On 9/5/11 2:41 PM, Ozren Lapcevic wrote:
</div></div><blockquote type="cite"><div><div></div><div class="h5">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) =~ "^<a>sip:00</a>") {<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 =~ "^<a>sip:00</a>") {<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) =~ "^<a>sip:00</a>") {<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>
}
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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</pre>
</blockquote><font color="#888888">
<br>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Kamailio Advanced Training, Oct 10-13, Berlin: <a href="http://asipto.com/u/kat" target="_blank">http://asipto.com/u/kat</a>
<a href="http://linkedin.com/in/miconda" target="_blank">http://linkedin.com/in/miconda</a> -- <a href="http://twitter.com/miconda" target="_blank">http://twitter.com/miconda</a></pre>
</font></div>
</blockquote></div><br>