Hello guys,
Is is possible to log the current (function) route?
Something like
route[WHATEVER]{
xlog("$curroute: Hello World\n");
}
and it would print:
WHATEVER: Hello World
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
Hello,
We're currently using the auth_db module and calling the function auth_check to check the Authorization header and authenticate the request.
We moved most of our code to in-memory lookups (using htable), since we do not want our Kamailio servers to be DB dependent.
Just to be sure, if we move the authentication to in-memory lookups we should use the auth module and call the pv_auth_check function instead right?
The passwords in our users table are all stored in MD5, so we basically just have to put this table in a htable, and call the pv_auth_check and pass it the hashed password and the flag 1?
Regards,
Grant
Hi,
I have two SIP servers, *A* and *B*, connected each other though a OPENVPN
tunnel. The server *B* needs to t_relay() every SIP message containing the
method MESSAGE to the server *A* but these messages never reach destination.
I have tested the tunnel connectivity and works fine. I wrote a Perl script
(located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying
to figure out what is happening but these messages are received by *A* and
processed correctly but when *B* does the same from Kamailio, it is never
received.
Here is the route part of kamailio.cfg in *B*:
Observation: ($rU == "1004") result is *true*
*
if(is_method("MESSAGE"))
{
if($rU == "1004")
{
xlog("L_INFO","En 1004");
rewritehost("10.8.0.1");
if (!t_relay())
xlog("L_INFO","MIO Error en t_relay");
t_reply("200", "Ok");
xlog("L_INFO","MIO despues rewrite");
exit;
}
.....*
The perl script that WORKS:
*$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2
From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796
To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>>
Call-ID: 19722852989(a)192.168.1.2
CSeq: 15773 MESSAGE
Contact: <sip:2002@**PublicIP-protected**:5060>
Max-Forwards: 29
User-Agent: DBL
Content-Type: text/plain
Content-Length: 34
+595981[protected]
hello from kamailio
';
use IO::Socket;
my $sock = IO::Socket::INET->new( Proto=>'udp',
PeerHost=>'192.168.2.102',
PeerPort=>'5060');
print "Sending msg $msg\n";
$sock->send($msg) or die "error sending $!\n";
*Please help!
Thanks in advance.
Carlos.
Hi,
I would like to know how many active calls are handled in kamailio server at a given point of time(concurrent calls).
In the few older post it was mentioned to use "dialog" module for these statistics.
In dialog module , there is a function called "active_dialogs" , which solves my requirement. But it is not clear how to use it. Can anybody help me out in this?
Thanks,
Vivek.
Hello,
Kamailio SIP Server v4.3.7 stable release is out.
This is a maintenance release of an old stable branch, 4.3, that
includes fixes since release of v4.3.6. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.3.x.
Important note: unless some major regression is discovered in the near
future, this is the latest release to be done from branch 4.3, which is
no longer to be officially maintained. Therefore it is strongly advised
to consider upgrading to 4.4.x or 5.0.x series.
For more details about version 4.3.7 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2017/06/kamailio-v4-3-7-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the latest stable branch is 5.0, at this moment with its latest
release v5.0.2. See more details about it at:
* https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/miconda - http://www.linkedin.com/in/miconda
You can view active dialogs with:
kamctl mi dlg_list
for less info
kamctl mi dlg_list | grep from-uri
or another header..
--------------------------------
Ángel Elena Medina _o)
craem(a)craem.net / \\
http://blog.craem.net _(___V
@craem_
--------------------------------
-----Mensaje original-----
De: vivek(a)advaitamtech.com
Enviado: Vie 23-06-2017 12:27
Asunto: [SR-Users] How to get the count of active calls handled in kamailio.
Para: sr-users(a)lists.sip-router.org;
> Hi,
>
>
> I would like to know how many active calls are handled in kamailio server at a
> given point of time(concurrent calls).
>
>
> In the few older post it was mentioned to use "dialog" module for these
> statistics.
>
>
> In dialog module , there is a function called "active_dialogs" , which solves
> my requirement. But it is not clear how to use it. Can anybody help me out in
> this?
>
>
> Thanks,
>
> Vivek.
>
>
> _______________________________________________
>
> Kamailio (SER) - Users Mailing List
>
> sr-users(a)lists.kamailio.org
>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
Hello,
I am planning to release Kamailio v4.3.7 very soon, likely tomorrow,
based on the last version of branch 4.3. This should mark the end of
official maintenance for branch 4.3, so if no major regression
discovered in following few weeks, it will be the last release in 4.3
series. If there is something you know it needs to be pushed in branch
4.3, write back to mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
Hi
I need to change the FROM header for every invite and replace the caller username with another data taken from mysql database.
Which is the best place in kamailio routes to do a uac_replace_from ?
It is correct to change the FROM header before the dialog creation, in “RELAY” route, just before the t_relay()?
Thank you
--
Emanuele Gambaro
---
email: emanuele.gambaro(a)pynlab.com
skype: sarbyn_work
OpenPGP Key: https://goo.gl/fdeVnI
Hi All,
Trying to work out a way to detect and re-route inbound calls which negotiate or contain t.38 SDP to answer/process faxes efficiently.
Plan is to put Kamailio in front of a quantity of FreeSwitch servers - most virtual, others physical.
Virtual servers will handle inbound faxes which negotiate t.38, and physical servers will answer ulaw/alaw faxes with mod_spandsp.
The bulk of inbound faxes negotiate t.38, but in order to scale our inbound system we need some way to work out which way to send the calls prior to the dispatcher.
Many thanks for your help in advance,
Tim