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

Create Talk Bots With XMPP Support On Google App Engine

Google App Engine where developers develop and launch web applications, on non other than Google's own infrastructure.

There are many apps that depends and developed using the Google App Engine. Google itself released apps like, translation bot that I constantly use on the Google talk (We wrote about the Translating chat bot a while back). I always wanted to have bot like that to help me and others but developing such a bot was not an easy task.
But now the tables have turned. Today Google App Engine team announced the release of an API that makes it very simple to build Talk bots on top of App Engine. I am looking forward to the clash of the bots!

Google App Engine SDK1.2.5 has more than just XMPP support, among usual bug fixes, tweaks, and API improvements, it now includes a Windows-based version of a useful tool, The Google App Engine Launcher!
The release notes have more information as well.

The whole technical features and more info is available from this fantastic article on Google App Engine Blog;
App Engine SDK 1.2.5 released for Python and Java, now with XMPP support

Jabber On Asterisk At Jabber Call Progress Branch

Asterisk Developer SVN server has a new branch now available for some testing, posted Asterisk Developer team member, bweschke. The new branch is at the following SVN URL:
http://svn.digium.com/asterisk/team/bweschke/originate_w_jabber/

This is based on the 1.4 branch and has had the res_jabber module that was in /trunk back-ported to it.

What does it do?

Phase 1 of the work here was to enable call progress messages to be sent via XMPP on Manager Originate actions and also within the Dial application when the appropriate channel variables are set.

What do the messages look like?

I did not get a chance to test it yet but according to the post, they are XML messages. Here is an example message so you get an idea of what the tree looks like.

progress
12345^^CALLID^^12345
PEER/1234567890: channel start

How do I make it work?

For this I have to direct you to post and the poster. It is described well enough to get you going by bweschke at Jabber call progress branch.

Jabber on...

tag: Asterisk, Jabber, XMPP, XML, , IP PBX, CALLID,