Hello all :-)
I have a question about how to get a time-stamps with milliseconds. To
create my own CDR's, i write to my databases timestamps, with a command like
this:
sql_query("ca", "UPDATE active_calls SET state = 'BILLING', end_time=$Ts
WHERE call_id='$ci'", "ra");
I use the "$Ts"-variable, but with this i can get only "full" seconds. But i
need milliseconds for my billing-system. Otherwise i have differences
between my CDR's and my carrier's one.
Example: A call has a duration of 0.2 seconds. In my CDR start- and endtime
is the same (=0 seconds), to my carrier a have 1 second to pay ;-)
Is there a possibility to get a time-stamp or time with milliseconds?
Cheers,
Marco
Marco Bungalski GmbH
Traversale 5
27283 Verden
Telefon: +49 4231 - 776 9999
Fax: +49 4231 - 776 9998
Mobil: +49 172 4204774
e-mail: Marco(a)Bungalski.de
web: www.bungalski.de
Geschäftsführender Gesellschafter: Marco Bungalski
Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586
Hello,
I will be in Berlin for few days (Jan 20-22), attending 3G+ and LTE
Forum. If anyone is around and want to meet, feel free to drop me an email.
Next event I will talk about Kamailio and SIP Router is FOSDEM in
Brussels -- more details will follow soon.
Cheers,
Daniel
--
Daniel-Constantin Mierla
* http://www.asipto.com/
Dear all,
I am using siptrace module and getting all the packets except the BYE.
when the call disconnected BYE packet is not received.
I am also getting the packets OK, REGISTER, INVITE two times in
sip_trace table.
Can anyone help me to sort out this problem??
--
Dinesh Gautam
!!
Hello all,
I know it might sound strange but, is there any way to deactivate that SER puts its Via on the SIP message
when forwarding it to other entity?
Thanks in advance,
Rebeca Martinez
ram schrieb:
> in that case we are going to loose registration right ?
not if you do database replication of "location" table
> and ongoing calls will be dropped right ?
No. only ongoing transactions may be affected.
regards
klaus
PS: please always Cc the mailing list
>
> Ram
>
>
>
> On Sat, Jan 16, 2010 at 2:58 AM, Klaus Darilion
> <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> wrote:
>
> Hi!
>
> For failover there are several possibilities. The simplest on, I
> guess the one you are referring to, is IP takeover.
>
> This is: 2 servers. On both is Kamailio and the database (e.g.
> mysql) installed. The SIP service uses a virtual IP address which is
> either configured at server 1 or server 2 (never on both). Kamailio
> process is only running on the server which has the virtual IP
> address provisioned.
>
> Then you need some other software to monitor the SIP service and
> trigger the failover from one server to the other server (e.g.
> either write some shell scripts or use existing software like
> hearbeat, ultramonkey, ...)
>
> Further - depending on your scenario - you might need some database
> replication (e.g. mysql cluster or mysql with master-master
> replication).
>
> regards
> klaus
>
> RAJNIKANT VANZA wrote:
>
> Hi,
>
> I want to implement kamailio server failover.
> so, anybody know about how to configure kamailio server failover?
>
> e.g. 2 kamailio severs run and when any one server fail then
> other server take over thats resources.
>
>
> --
> Best Regards,
>
> Rajnikant Vanza
>
>
> ------------------------------------------------------------------------
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org>
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org>
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
Hi,
I have a problem that I don't understand at all (yet another one ;-)).
One client has Asterisk (like many others) from where he sends calls
to our Kamailio. It worked fine, until it suddenly stopped for no
apparent reason on Friday. No changes in configuration on his or our
side.
We can ping his server just fine, and vice versa.
On any INVITE he sends, he gets no, zero, zip response from Kamailio
on our server. I wanted to figure out why, so I put an 'ngrep' on port
5060 for his IP on our Kamailio server, and I got...nothing. No
communication at all on this port containing this IP. Why? I have no
clue. The ping should show that we can communicate, and a 'ngrep' on
his server shows that he sends plenty of INVITEs, but gets absolute
zero response from our Kamailio box.
Any ideas where to look?
Thanks a lot in advance!
Hi,
the sources of siremis are in sourceforge.net svn. If any of you has
developed (plan to develop) extensions, contact me to grant devel
access. Also, patches should be sent against latest svn trunk (no
branches so far):
http://sourceforge.net/projects/siremis/
The svn repo does not store the openbiz core framework -- you have to
get it from siremis-1.0.0.tgz (the openbiz dir) or download it from
openbiz project and apply the patch from misc directory (note that you
also have to download zend and others according to openbiz guides).
Once you fetch siremis svn and add openbiz, run make to create the
symlinks required for web dirs and you are set.
Regards,
Ramona
Hi,
I have just released SIREMIS 1.0.0 - web admin interface for Kamailio
(OpenSER) 3.0.0 and SIP Router.
Among new features:
* presence service management – watchers, active watchers and presentity
* sip trace view – access sip_trace module records, highlight headers
and group by call-id
* user location statistics charts – online user agents, supported SIP
methods, contacts and nat related charts
* update of the core components to latest stable versions
* update to work with latest Kamailio (OpenSER) v3.0.0
* patch to work with older Kamailio (OpenSER) v1.5.x
* time-based X-axis for charts print labels as HH:MM
* charts types supported: line, line dot and area
Some screenshots specific for this version:
http://www.asipto.com/gallery/v/siremis/siremis_22.png.htmlhttp://www.asipto.com/gallery/v/siremis/siremis_23.png.htmlhttp://www.asipto.com/gallery/v/siremis/siremis_24.png.html
Download and installation steps:
http://siremis.asipto.com/install/
More screenshots:
http://www.asipto.com/gallery/v/siremis/
Demo site (it works on a database with random data, username: admin,
password: admin):
http://siremis.asipto.com/demo/
Regards,
Ramona
Hello.
I have the next situation with kamailio 3.0.0
In my config file I have something like this :
route
If INVITE then route(AUTH_REQUEST)
. . .
route[AUTH_REQUEST]
. . .
if (client_nat_test("3")) {
append_hf("P-hint:
route(AUTH_REQUEST)|setflag7,forcerport,fix_contact\r\n");
if (setbflag("7")) {
xlog("L_INFO","[$ci]:[AUTH_REQUEST] :
setbflag(7)");
}
force_rport();
fix_contact();
xlog("L_INFO","[$ci]:[AUTH_REQUEST] :
CLIENT_NAT_TEST-3");
};
. . .
route(IN_TO_IN)
route[IN_TO_IN]
. . .
route(LOCAL_CALL)
. . .
route[LOCAL_CALL]
. . .
if (isbflagset("7") ) {
xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET begin");
} else {
xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NO SET
begin");
}
if ( !lookup("location") ) {
if ( avp_check("$avp(s:4)", "eq/s:1/g") ) {
if( !avp_pushto("$br", "$avp(s:5)/g") ) {
sl_send_reply("403","Forbidden - Forward
Number NULL");
exit;
};
prefix("222");
route(TO_VOICEMAIL_NODISP);
exit;
};
if ( avp_check("$avp(s:3)", "eq/s:1/g") ) {
xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm -
From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL ON\n");
route(TO_VOICEMAIL_NODISP);
exit;
} else {
sl_send_reply("480", "Usuario fuera de
Servicio");
xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm -
From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL OFF\n");
exit;
};
};
if (isbflagset("7") ) {
xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET end");
} else {
xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NOT SET
end");
}
. . .
For some reason in the route[LOCAL_CALL] the flagb "7" is changed from
"SET" to "NOT SET". According to my debugs this unset is made in the
"lookup(location)" part.
But I don't understand why.
Can someone help me here?
Thanks in advance,
Ricardo Martinez.-
I'm trying to append an RPID header on my SER 0.9.7-pre4 system. The config has not changed and this use to work in SER 0.9.7-pre3. The header does not get appended and I cannot see why. The use_rpid parameter or the auth_db module does not seem to exist in this release. Is there a different way to append the header?
Thanks,Steve