New subject: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration

29 Dec 2010

Apparently the process of the installation did not setup even a default
sip.conf file, and there were some other issues with the Kamailio.cfg file.
Most of the issues were resolved by one of the other users in the group, and
we were able to get the system running.

Thanks everyone for your assistance.

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Today's Topics:

   1. Re: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration
      using Asterisk Database (Klaus Darilion)
   2. Load Balancing (Tim King)
   3. Re: Load Balancing (Alex Balashov)
   4. fifo dp_translate (Borin)
   5. Re: some issues with database initialization	(berkeley/txt)
      (Noa Resare)
   6. Re: some issues with database initialization	(berkeley/txt)
      (Daniel-Constantin Mierla)

----------------------------------------------------------------------

Message: 1
Date: Wed, 29 Dec 2010 15:17:41 +0100
From: Klaus Darilion &lt;klaus.mailinglists(a)pernau.at&gt;
Subject: Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime
	Integration using Asterisk Database
To: sr-users(a)lists.sip-router.org
Message-ID: &lt;4D1B4305.30504(a)pernau.at&gt;
Content-Type: text/plain; charset=windows-1252; format=flowed

On 28.12.2010 22:07, Kurt Mullen wrote:
...
  I am on my 20^th (I?m not kidding) attempt to
successfully complete 
 this tutorial.

 I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & 
 Asterisk on the same server as in the tutorial.

 I have two SIP clients registered, but they are not able to call each other.
...

 No one answered my last two posts, so I hope someone can help this time. 
If you can describe the problem (why can't they call each other) we can help
you.

So, first you have to find out what is the problem. Use a packet sniffer
(wireshark, ngrep) to capture the SIP traffic and analyze it (yes, if you
want to use Kamailio you have to know at least the basics of SIP) for the
problems.

regards
klaus

------------------------------

Message: 2
Date: Wed, 29 Dec 2010 09:31:28 -0500
From: Tim King &lt;tim(a)compnetwork.net&gt;
Subject: [SR-Users] Load Balancing
To: sr-users(a)lists.sip-router.org
Message-ID:
	&lt;AANLkTindx_r-RmTJSjTnpfdmLX8Nespzs_M9O0f6y385(a)mail.gmail.com&gt;
Content-Type: text/plain; charset="iso-8859-1"

I have read many examples and it seems I have found several pieces of how to
do this but no real example of how to achieve this.

I followed this setup guide:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc

Performed this Kamailio  install:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git

I have found the page for the dispatcher module here:
http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html

I also am curious if this dispatcher setup can be driven solely from the
dispatcher mysql table or if the dispatcher.list file is required.

Can someone point me towards the proper documentation that truly show how to
configure these features.

*SIP Routing Capabilities*

   -
      - NAT traversal support for SIP and RTP traffic
      - load balancing with many distribution algorithms and failover
      support
      - flexible least cost routing
      - routing failover
      - replication for High Availability (HA)

* Load Based Load Balancing
** -------------------------
**
** +----------+
** | Kamailio |
** +----------+
**   |
**   |       +--------------+
**   |-------| FreeSWITCH 1 |  [Calls 1,3,7]
**   |       +--------------+
**   |
**   |       +--------------+
**   |-------| FreeSWITCH 2 |  [Calls 2,6]
**           +--------------+
*

Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration