[SR-Users] Accounting only the 2nd branch of missed serial forked call
Daniel-Constantin Mierla
miconda at gmail.com
Wed Sep 7 13:05:47 CEST 2011
Hello,
On 9/7/11 11:25 AM, Ozren Lapcevic wrote:
> Hi Daniel,
>
> thanks for the quick fix and reply.
>
> 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
> (http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git)
> to get t_flush flags() and your patch or is there a workaround to
> apply them to my 3.1.4 branch?
did you install 3.1.4 from tarball/packages or is it from git branch
3.1? If later, then you can do:
git pull origin
git cherry-pick -x 83620cb7cd14ee3b509eef72d99337567f53967f
git cherry-pick -x c589ca35b2aa3097a3c9e2a5a050514337300c05
then recompile/install. First cherry-pick brings the t_flush_flags, the
second auto-update of the flags after branch/failure route.
If you installed from packages, then you would need to repackage
yourself after patching. The patches are available at commit url, for
example:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c589ca35b2aa3097a3c9e2a5a050514337300c05
There you find at top of the page a link named 'patch' that you can use
with git tools to apply or extract the diff-patch part and apply with patch.
Cheers,
Daniel
>
> Cheers
> Ozren
>
>
> On Tue, Sep 6, 2011 at 2:18 PM, Daniel-Constantin Mierla
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
> Hello,
>
> 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:
>
> http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c589ca35b2aa3097a3c9e2a5a050514337300c05
>
> Actually, reporting if all goes fine with this patch, will help in
> backporting it to 3.1 branch.
>
> Thanks,
> Daniel
>
>
> On 9/5/11 2:41 PM, Ozren Lapcevic wrote:
>> Hi,
>>
>> I'm having some problems accounting missed serial forked calls to
>> mysql database.
>>
>> 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.
>>
>> For example, I have two users: oz at abc.hr <mailto:oz at abc.hr> and
>> pero at abc.hr <mailto:pero at abc.hr>. pero at abc.hr
>> <mailto:pero at abc.hr> also has set telephone number as alternative
>> number if he is not reachable at sip:pero at abc.hr
>> <mailto:sip%3Apero at abc.hr>. Moreover, pero at abc.hr
>> <mailto:pero at abc.hr> has voicemail turned on. When oz at abc.hr
>> <mailto:oz at abc.hr> calls pero at abc.hr <mailto:pero at abc.hr>, first
>> pero at abc.hr <mailto:pero at abc.hr>'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.
>>
>> There are two interesting scenarios accounting-wise which can
>> happened:
>> 1. oz at abc.hr <mailto:oz at abc.hr> calls pero at abc.hr
>> <mailto:pero at abc.hr>, there are no answers and call is forked to
>> voicemail.
>> 2. oz at abc.hr <mailto:oz at abc.hr> calls pero at abc.hr
>> <mailto:pero at abc.hr>, there is no answer at SIP client, but pero
>> answers call at telephone.
>>
>> When scenario 1 happens, I want to have only one log (row) in
>> missed_calls table.
>>
>> When scenario 2 happens, I don't want to have a log in
>> missed_calls table.
>>
>> To accomplish this,*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*. 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 $ru = "something" to rewrite URI prior to forking request.
>>
>> I can easily set up logging of every call (two missed calls for
>> serially forked call to two locations) by setting FLT_ACCMISSED
>> flag for each INVITE. I can set up logging of every call's 1st
>> branch, by reseting FLT_ACCMISSED flag when handling 2nd branch
>> of the call. Interestingly, logging of only the 2nd branch of the
>> serial forked call works when there is no forking to voicemail!
>>
>> Any ideas how to solve this problem?
>>
>> Bellow are important parts of my config file. I'm running
>> kamailio 3.1.4.
>>
>> Cheers
>> Ozren
>>
>>
>> # ----- acc params -----
>> /* what special events should be accounted ? */
>> modparam("acc", "early_media", 0)
>> modparam("acc", "report_ack", 1)
>> modparam("acc", "report_cancels", 0)
>> modparam("acc", "detect_direction", 0)
>> /* account triggers (flags) */
>> modparam("acc", "log_flag", FLT_ACC)
>> modparam("acc", "log_missed_flag", FLT_ACCMISSED)
>> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
>> modparam("acc",
>> "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
>> /* enhanced DB accounting */
>> #!ifdef WITH_ACCDB
>> modparam("acc", "db_flag", FLT_ACC)
>> modparam("acc", "db_missed_flag", FLT_ACCMISSED)
>> modparam("acc", "db_url", DBURL)
>> modparam("acc", "db_extra",
>> "src_user=$fU;src_domain=$fd;dst_user=$tU;dst_domain=$td;src_ip=$si")
>> #!endif
>>
>> ...
>>
>>
>> # Main SIP request routing logic
>> # - processing of any incoming SIP request starts with this route
>> route {
>>
>> # per request initial checks
>> route(REQINIT);
>>
>> if (src_ip != ****) {
>> # NAT detection
>> route(NAT);
>> }
>>
>> # handle requests within SIP dialogs
>> route(WITHINDLG);
>>
>> ### only initial requests (no To tag)
>>
>> # CANCEL processing
>> if (is_method("CANCEL"))
>> {
>> if (t_check_trans())
>> t_relay();
>> exit;
>> }
>>
>> t_check_trans();
>>
>> # authentication
>> route(AUTH);
>>
>> # record routing for dialog forming requests (in case
>> they are routed)
>> # - remove preloaded route headers
>> remove_hf("Route");
>> if (is_method("INVITE|SUBSCRIBE"))
>> record_route();
>>
>> # account only INVITEs
>> if (is_method("INVITE"))
>> {
>> setflag(FLT_ACC); # do accounting
>> }
>>
>> # dispatch requests to foreign domains
>> route(SIPOUT);
>>
>> ### requests for my local domains
>>
>> # handle presence related requests
>> route(PRESENCE);
>>
>> # handle registrations
>> route(REGISTRAR);
>>
>> if ($rU==$null)
>> {
>> # request with no Username in RURI
>> sl_send_reply("484","Address Incomplete");
>> exit;
>> }
>>
>> # dispatch destinations to PSTN
>> route(PSTN);
>>
>> if ( is_method("INVITE") ) {
>> route(DBALIASES);
>> #check for user defined forking priorities and timers
>> route(FORK);
>> }
>>
>> # user location service
>> route(LOCATION);
>>
>> route(RELAY);
>> }
>>
>>
>>
>> #check for user defined forking priorities and timers
>> route[FORK]{
>> sql_query("con", "select * from usr_pref_custom where
>> uuid='$tu'", "pref");
>>
>> $avp(uuid)=$dbr(pref=>[0,0]);
>> $avp(email)=$dbr(pref=>[0,1]);
>> $avp(prio1)=$dbr(pref=>[0,2]);
>> $avp(prio2)=$dbr(pref=>[0,3]);
>> $avp(timer1)=$dbr(pref=>[0,5]);
>> $avp(timer2)=$dbr(pref=>[0,6]);
>>
>> if (strlen($avp(prio1))>5) {
>>
>> # user has multiple contacts, do serial forking
>> setflag(FLT_USRPREF);
>>
>> # set counter
>> if (!$avp(prio)) {
>> $avp(prio) = 1;
>> }
>>
>> # overwrite request URI with highest priority contact
>> if ($avp(prio1) =~ "^sip:00") {
>> $ru = $avp(prio1) + "@host";
>> xlog("L_INFO","PRIO 1 is tel number, RURI
>> set: $ru");
>> }
>> else {
>> $ru = $avp(prio1);
>> xlog("L_INFO","PRIO 1 is SIP URI, RURI
>> set: $ru");
>> }
>> }
>> }
>>
>>
>> route[RELAY] {
>> #!ifdef WITH_NAT
>> if (check_route_param("nat=yes")) {
>> setbflag(FLB_NATB);
>> }
>> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
>> route(RTPPROXY);
>> }
>> #!endif
>>
>> /* example how to enable some additional event routes */
>> if (is_method("INVITE")) {
>>
>> t_on_reply("REPLY_ONE");
>> t_on_failure("FAIL_ONE");
>>
>> #if users have priorities set, use FAIL_FORK
>> failure route
>> if ( isflagset(FLT_USRPREF) ) {
>> t_on_failure("FAIL_FORK");
>> }
>> }
>>
>> if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm
>> $ru, ACCMISSED FLAG IS SET");
>> else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT
>> SET");
>> if (!t_relay()) {
>> sl_reply_error();
>> }
>> exit;
>> }
>>
>>
>> # Handle requests within SIP dialogs
>> route[WITHINDLG] {
>> if (has_totag()) {
>> # sequential request withing a dialog should
>> # take the path determined by record-routing
>> if (loose_route()) {
>> xlog("L_INFO","WITHINDLG, loose_route()");
>> if (is_method("BYE")) {
>> xlog("L_INFO","WITHINDLG, BYE, DO
>> ACCOUNTING");
>> setflag(FLT_ACC); # do accounting ...
>> setflag(FLT_ACCFAILED); # ...
>> even if the transaction fails
>> }
>> route(RELAY);
>> } else {
>> if (is_method("SUBSCRIBE") && uri ==
>> myself) {
>> # in-dialog subscribe requests
>> route(PRESENCE);
>> exit;
>> }
>> if ( is_method("ACK") ) {
>> if ( t_check_trans() ) {
>> # no loose-route, but
>> stateful ACK;
>> # must be an ACK after a 487
>> # or e.g. 404 from
>> upstream server
>> t_relay();
>> exit;
>> } else {
>> # ACK without matching
>> transaction ... ignore and discard
>> exit;
>> }
>> }
>> sl_send_reply("404","Not here");
>> }
>> exit;
>> }
>> }
>>
>>
>> # USER location service
>> route[LOCATION] {
>>
>> #skip if $ru is telephone number
>> if ($ru =~ "^sip:00") {
>> xlog("L_INFO","SKIP lookup...");
>> }
>> else {
>> if (!lookup("location")) {
>> switch ($rc) {
>> case -1:
>> case -3:
>> t_newtran();
>> t_reply("404", "Not Found");
>> exit;
>> case -2:
>> sl_send_reply("405",
>> "Method Not Allowed");
>> exit;
>> }
>> }
>> }
>>
>> # when routing via usrloc, log the missed calls also, but
>> only if user doesn't have prios set
>> if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) {
>> setflag(FLT_ACCMISSED);
>> }
>> }
>>
>>
>> # Failure route for forked calls
>> failure_route[FAIL_FORK] {
>> #!ifdef WITH_NAT
>> if (is_method("INVITE") && (isbflagset(FLB_NATB) ||
>> isflagset(FLT_NATS))) {
>> unforce_rtp_proxy();
>> }
>> #!endif
>>
>> if ($avp(prio) >= 1) {
>> $avp(prio) = $avp(prio) + 1;
>>
>> # handle 2nd branch
>> if ( ($avp(prio) == 2) && (
>> isflagset(FLT_USRPREF) )) {
>> t_on_failure("FAIL_FORK");
>>
>> if ($avp(prio2) =~ "^sip:00") {
>> xlog("L_INFO","FAIL FORK, PRIO 2
>> is tel number");
>> $ru = $avp(prio2) + "@host";
>> }
>> else {
>> xlog("L_INFO","FAIL FORK, PRIO 2
>> is SIP URI");
>> $ru = $avp(prio2);
>> route(LOCATION);
>> }
>> setflag(FLT_ACCMISSED);
>> }
>>
>> else {
>> $avp(prio) = 0;
>> $ru = $(avp(uuid));
>> rewritehostport("host:port");
>> xlog("L_INFO","FAIL FORK, VOICEMAIL
>> email:$avp(email), ru:$ru, br: $br");
>> append_hf("P-App-Name: voicemail\r\n");
>> append_hf("P-App-Param:
>> Email-Address=$avp(email)\r\n");
>> }
>> route(RELAY);
>> }
>>
>> if (t_is_canceled()) {
>> exit;
>> }
>> }
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla --http://www.asipto.com
> Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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