[SR-Users] Accounting only the 2nd branch of missed serial forked call
Ozren Lapcevic
ozren.lapcevic at gmail.com
Wed Sep 7 14:20:17 CEST 2011
Hi,
I've previously installed kamailio from git branch 3.1. Now, I've manged to
git cherry-pick your patch, but got "fatal: Could not find
83620cb7cd14ee3b509eef72d99337567f53967f" when tried to get t_flush_flags().
I've double-checked commit and found it here:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=83620cb7cd14ee3b509eef72d99337567f53967f.
I don't know why I can't cherry-pick it.
Your patch alone, without t_flush_flags(), doesn't change anything in my
scenario, there is still no logging of 2nd branch.
Cheers
Ozren
On Wed, Sep 7, 2011 at 1:05 PM, Daniel-Constantin Mierla
<miconda at 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 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 and pero at abc.hr. pero at abc.hralso has set telephone number as alternative number if he is not reachable
>> at sip:pero at abc.hr. Moreover, pero at abc.hr has voicemail turned on. When
>> oz at abc.hr calls pero at abc.hr, first 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 calls pero at abc.hr, there are no answers and call is forked
>> to voicemail.
>> 2. oz at abc.hr calls 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 listsr-users at lists.sip-router.orghttp://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/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://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/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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