Hello Daniel,
Yes, dlg_manage() is being called in the BYE, yet it sometimes still occurs
that dialogs aren't being cleared.
It is critical for us the dialogs are being cleared as we have an application
which heavily relies on this.
Kind regards,
Grant Bagdasarian
From: Daniel-Constantin Mier
Hi Matt,
I don't know about Siremis, but I think nice tutorials are:
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
and
http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
They might probably not answer all your questions, but in my opinion
they show quite nice h
Before using my suggestion below, please consider first that I am a "dummy
programmer" and then, that I am using Ubuntu server 12.04 LTS 64 bits. If you
are using another distro or version, you may need to adapt the upstart script.
Also, I am not sure if this is the right way of doing it, but i
Seems like Kamailio is configured to require a client certificate, but
the client doesn't have one.
klaus
Am 15.11.2012 15:04, schrieb Denis:
Thank you, it worked!
I just added listen=tcp:IP:5060 so it probably needs to initialize tcp
separately from tls )
Thanks a lot!
P.S.
now I am havin
15 nov 2012 kl. 11:58 skrev Christophe ROY :
> Hi everyone
>
> I'm trying to integrate Asterisk with Kamailio for voicemail.
> I tried to follow this tutorial:
> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
> BUT:
>
> - I had to adapt it because I use LDAP authen
15 nov 2012 kl. 16:53 skrev Krishna Kurapati :
> Hi,
>
> Can Kamailio act as SMTP server to receive emails and forward those to SIP
> users as MESSAGEs?
>
> It would be nice if postfix or some other rich SMTP server could function as
> module of kamailio using all the services provided by kam
Hi,
Can Kamailio act as SMTP server to receive emails and forward those to SIP
users as MESSAGEs?
It would be nice if postfix or some other rich SMTP server could function
as module of kamailio using all the services provided by kamailio scripting.
Krish Kura
On Thu, 15 Nov 2012 15:00:24 +0100 Pedro Antonio Vico Solano
wrote:
> We've done more tests with both of your patches and it seems to work
> fine.
>
> You can upload it to GIT.
Done.
Commit ecf95eb Wed Oct 17 09:00:14 fixes the mem leak.
Commit 0920586 Thu Nov 15 16:11:41 fixes the computed fi
Thank you, it worked!
I just added listen=tcp:IP:5060 so it probably needs to initialize tcp
separately from tls )
Thanks a lot!
P.S.
now I am having another errors though while connecting to tls port but I
believe it is certificates problems:
$ openssl s_client -connect IP:5061 -tls1 -CAfil
Hello Timo.
We've done more tests with both of your
patches and it seems to work fine.
You can upload it to GIT.
Thanks a lot !!!
Regards
De:
Timo Teras
Para:
Timo Teras
cc:
"SIP Router -
Kamailio \(OpenSER\) and SIP Express Router \(SER\) -
Users Mailing List"
Copy and paste typo, overlapping port use:
listen=tcp:127.0.0.1:5060
Cheers,
Daniel
On 11/15/12 7:54 AM, Denis wrote:
Thanks for looking at that, Daniel.
If I start all together with tls: and tcp: (both lines order) then I
see this:
/opt/kamailio/sbin/kamailio[1008]: ERROR: [tcp_main.c:2
My configuration is
SIP client <-> Kamailio <-> Asterisk <-> Kamailio <-> SIP client
You can start from http://www.kamailio.org/wiki/start#tutorials
Dialing from Asterisk looks like:
SIP/KamailioIP/NumberToDial,gwWL(3307996)
In Kamailio if you put IP of Asterisk servers in correct tables, you
can
Thanks for looking at that, Daniel.
If I start all together with tls: and tcp: (both lines order) then I see
this:
/opt/kamailio/sbin/kamailio[1008]: ERROR: [tcp_main.c:2918]:
ERROR: tcp_init: bind(9, 0x7f3fa8eb7d64, 16) on IP_ADDRESS:5061 :
Address already in use
/opt/kamailio/sbin/kamai
Reviewing the previous email, I probably spotted the issues. You said
you added:
listen=tls:IP.ADDRESS:5061
that forces Kamailio to listen only on tls. But tls is on top of tcp, so
add:
listen=tcp:127.0.0.1:5061
It was reported couple of days ago, I had no time to look at it yet,
traveling
Only during kamailio start:
...
/opt/kamailio/sbin/kamailio[752]: INFO: rtpproxy [rtpproxy.c:1413]: rtp
proxy found, support for it enabled
/opt/kamailio/sbin/kamailio[759]: INFO: ctl [io_listener.c:225]:
io_listen_loop: using epoll_lt io watch method (config)
as soon as I send a request on
Hello,
On 11/15/12 4:52 AM, Denis wrote:
[...]
/opt/kamailio/sbin/kamailio[30278]: ALERT: [main.c:785]: child
process 30293 exited by a signal 11
Before this line, do you have any other error messages printed by pid 30278?
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
h
Anybody got a good tutorial or starter project for Kamailio?
We have Siremis 3 installed, but there is 0 help on that?
Trying to solve 2 scenarios:
-asterisk to softphone through kamailio
-softphone to asterisk through kamailio.
I ultimately want to use Kamailio as an edge router
Any help is app
- can you enable core dumping per process, just to be sure this core
is not caused by a kill or children processes?
I enabled core dump per process and can see that the only 1
core.pidnumber file generated with kamailio PID
___
SIP Express Router
Hi Daniel,
Thanks for your reply.
- what is the ssl lib version?
OpenSSL 0.9.8o
- it is happening all the time you try?
as soon as the phone send package on port 5061 the server crashes.
openssl s_client -debug -connect HOSTNAME:5061 -tls1
crashes Kamailio as well.
- what are the log mess
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