[SR-Users] Kamailio v4.3 and Freeswitch Stable Latest

Barış Şekerciler baris.sekerciler at outlook.com
Fri Jan 29 11:40:00 CET 2016


Hello again,I authanticated with Kamailio and communicate with IP phones succesfully.But depends on this tutorial: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbcWhen I tried to call "44102" I should go to voicemail box of user 102 but phone says "not found" and I think Kamailio could not route to Freeswitch.
BTW, what kind of logs do you want? Level? How can I send these are?
Regards.
From: baris.sekerciler at outlook.com
Date: Fri, 29 Jan 2016 11:37:54 +0200
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio v4.3 and Freeswitch Stable Latest

Hello Jurjis, thanks for your answer.
I did not try default IVR with Freeswitch. But I will try and write.
Regards.
On 29 Jan 2016, at 11:32, Jurijs Ivolga <jurij.ivo at gmail.com> wrote:Hi Baris,

Are you able to dial into Freeswitch default IVR(5000 ext)?

It will helps if you send logs from Kamailio and freeswitch.

With kind regards,

2016-01-29 11:14 GMT+02:00 Barış Şekerciler <baris.sekerciler at outlook.com>:
Thanks for answer Phill. Actually I did not apply any different configuration than digetics in Kamailio Wiki. (I installed Kamailio latest stable from Deb Jessie repos)After that, I applied these are: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc and I adapted to Kamailio 4.3 (at least I tried)There is nothing to different from this link.
So, I work on VmWare.Kamailio Server IP:       192.168.2.98FreeSwitch Server IP:   192.168.2.131




On 29 Jan 2016, at 10:38, Phil Lavin <phil.lavin at synety.com> wrote:
You'll need to share your Kamailio config and some details of your network setup (specifically if NAT is involved).


Phil

-----Original Message-----
From: sr-users [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Baris Sekerciler
Sent: 29 January 2016 08:35
To: sr-users at lists.sip-router.org
Subject: [SR-Users] Kamailio v4.3 and Freeswitch Stable Latest

Hello everyone. I'm a student and I'm working on project which about SIP communication and PBXs with IP phones.

So, my first test results were OK with JITSI. But I think I need Freeswitch for working with IP phones. In a word, I need configure an integration Kamailio and Freeswitch.

First, I setup Debian 8.3 Jessie and installed Kamailio on this my local server, after that I installed another Debian 8.3 Jessie (from same image) as my second server and it was going to run Freeswitch on itself.
Then I tried to send my communication requests to my Freeswitch server. I made some tests, my Jitsi accounts which are set in Kamailio server were done. I've read Kamailio 3.2 and Freeswitch integration from the Kamailio wiki and I applied exactly the same thing but I did not work.
When I tried to reach another IP phone from my IP phone (I registered them) phone always say "NOT FOUND". I'm sure I did my configuration properly.

So, how can I figure it out these things? How can I understand, does Kamailio routes to Freeswitch to my phones?

Best regards.
Thanks.
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