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AstLinux

AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as FreeSWITCH are also available. AstLinux has many unique features that make it ideal for embedded and commercial Asterisk based solutions.

AstLinux contains powerful networking features including:

Supported platforms include:

The following telephony hardware cards are supported:

The AstLinux Team recently announced the release of the 1.0.1 version. This version is available with either Asterisk 1.4.43 or Asterisk 1.8.8.3. A full changelog and upgrade (or new install) instructions are available on their website. Please follow the upgrade instructions carefully when upgrading from a release prior to 1.0.

http://www.astlinux.org

As always, please report any issues (and comments) to the AstLinux mailing list on Sourceforge. (link available at the above website).

H.323 Video Conferencing with GNU Gatekeeper

H.323 Video Conferencing with GNU Gatekeeper has been improved with the new released announced by Jan Willamowius, Founder of the GNU Gatekeeper Project.

A new major release of the GNu Gatekeeper has been anounced. This new version “has many new features that will allow even your legacy endpoints to move to a new age of H.323 audio and video communications.”

New features:
– full traversal zone support (gatekeeper-to-gatekeeper H.460.18/.19)
Now you can place one GnuGk behind a firewall and let it tunnel out the calls for all other devices behind the firewall eg. to a VCS or to another GnuGk.
This was probably the most request feature in the past.

– full IPv6 support (incl. IPv4-IPv6 proxying)
With the proxy function, you can let GnuGk manage a network of IPv6 endpoints and connect them to the IPv4 network or make legacy endpoints reachable for IPv6 calls.

RTP multiplexing (all calls to and from devices supporting H.460.19 will only use 2 sockets total)

– rewrite destination IPs into aliases

– ENUM, SRV and RDS routing policies extended for LRQs, in case the calling gatekeeper isn’t able to do this

– notifications when GnuGk opens listen ports
This allows you to update firewall rules on the fly, so you only have the minimum amount of ports open.

– improved H.235 password authentication with neighbors

– massive performance improvement when (re-)loading large numbers of GW rewrites

– interop fixes for Polycom m100 and Sorenson endpoints

– fixes in the underlying libraries so *BSD systems can get the latest GnuGk features

– a few bug fixes

The project provides executables for Linux (32 and 64 bit), Windows, MacOS X, FreeBSD, OpenBSD, NetBSD and Solaris.

You can download the new version at
http://www.gnugk.org/h323download.html

 

Enjoy the new release of such a great software!

Speech recognition in Asterisk using Google Voice API

I’m exited to announce that Lefteris Zafiris has written an agi script that uses Google Voice API for voice recognition.

As the author says, “the script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable.”

More info and dialplan examples can be found in the README file:
https://raw.github.com/zaf/asterisk-speech-recog/master/README

The script is available here:
https://github.com/zaf/asterisk-speech-recog

The author reports that the code is still young and not roughly tested so comments, suggestions and bug reports are more than welcome.

Enjoy this code jewel and please provide feedback/comments to the author.

Asterisk 1.6.2.22 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013:

* The sample file listed *two* values for the ‘nat’ option as being the default. Only ‘yes’ is the default.

* The warning about having differing ‘nat’ settings confusingly referred to both peers and users.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22

Thank you for your continued support of Asterisk!

 

AstLinux 1.0.0 release

The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This release includes significant changes and improvements over past releases. Specific upgrade and new installation instructions are available at: http://www.astlinux.org

Some of the highlights include:

* Using eglibc instead of uClibc. This allows binary compatibility with add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc).
* Newer Kernel which better supports newer hardware
* Support for Jabber/Gtalk
* Removed mISDN support (the zaphfc DAHDI driver is included for single port ISDN cards)

A full changelog is available on the release pages. We provide versions with Asterisk 1.8 and 1.4.

Because this is a major version change, there are some special considerations when upgrading. Please read the instructions very carefully to ensure no step is skipped.

http://doc.astlinux.org/userdoc:upgrade-0.7

Please report any issues with the release back to the AstLinux mailing list.

Enjoy,

The AstLinux Team

Asterisk 10.0.0 Is Released!

The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

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:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

With the release of the Asterisk 10 branch, the preceding ‘1.’ has been removed from the version number per the blog post available at

http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

The release of Asterisk 10 would not have been possible without the support and contributions of the community.

You can find an overview of the work involved with the 10.0.0 release in the summary:

http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt

A short list of available 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://svn.asterisk.org/svn/asterisk/branches/10/CHANGES

Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file,  which is located at:

http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt

Thank you for your continued support of Asterisk!