Asterisk 10 Beta 1 From New Asterisk 10 Branch Is Ready For Downloading And Testing

http://snapvoip.blogspot.com/
The Asterisk Development Team announced the availability of the first beta release of Asterisk10, Asterisk 10.0.0-beta1. The release is available download and consumption at Asterisk download site.

As we mentioned before, Digium and Asterisk.org has dropped "1." from the Asterisk version numbers and the new Asterisk 10 branch, will continue to march forward.
Of course Asterisk team is requesting your participation in testing, we are testing it with PIAF-RED;

All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases. More information

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk versions page:

A short list of included features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog

OpenSIPS 1.6 eBootcamp, Get Ready For A Torrent Of SIP Technology

OpenSIPS 1.6 eBootcamp http://snapvoip.blogspot.com/

OpenSIPS is holding a eBootcamp on OpenSIPS 1.6. This bootcamp is unique in itself because it is held over the Internet for over 7 weeks. Participants will learn from downloading installing and configuring and on to administration of OpenSIPS. It will start on September 19th 2011. The ebootcamp will accept late registrations up until 12th September 2011. So you better hurry. Contact bootcamp@opensips.org for registration and more information

The users will learn how to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.

This is basically the same OpenSIPS Bootcamp but help over a broadband connection over a longer period, seven weeks to be exact.

"The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials."

The course of learning will cover key objectives;

Install OpenSIPS on a Linux Machine

Routing basics and the default configuration

OpenSIPS authentication using MySQL and Memcache

Install OpenSIPS control Panel.

Connect to the PSTN using Dialplan and Dynamic Routing

Voicemail integration using Call Forward and AVPs

Implement a presence agent

Understand important aspects of load balancing and high availability

Implement SIP NAT traversal using RTPProxy

Account Calls to MySQL

How to use test and monitoring tools to check your configuration

OpenSIPS