On Wed, Sep 7, 2011 at 1:05 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:

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@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@abc.hr and pero@abc.hr. pero@abc.hr also has set telephone number as alternative number if he is not reachable at sip:pero@abc.hr. Moreover, pero@abc.hr has voicemail turned on. When oz@abc.hr calls pero@abc.hr, first pero@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@abc.hr calls pero@abc.hr, there are no answers and call is forked to voicemail.
2. oz@abc.hr calls pero@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;
        }
}
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sr-users@lists.sip-router.org
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-- 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@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