The news of Digium IP Phones have got a big number of people giving their opinion on the subject. Some of the inquiries are about the manufacturing process, others even question the very reasons for Digium to launch phones (after all, they are entering to a very tight and busy phone market).

The fact that Digium itself is manufacturing and designing the phones brings good news for us, developers. The Digium’s team was ordered to create the easiest phone to install, integrate, provision and use. After the general availability date of these phones comes (April 2012), we will see if the engineers delivered on that goal.

Advantages:

• The new phones are easier to auto-discover and configure them.
•  The application an the phone now will be more integrated as the phones have been thought in a way that they can access all sorts of user status and system information.
• Custom developer’s application will now be able to run directly in the phone thanks to an application engine that has been included.

In those places with good broadband connection service available, like O2, this opens a new realm of possibilities. Now the tools for a complete VoIP solution have been provided with the introduction of this Astersisk-specifical oriented phones.

Could you even imagine what could a community of more than 80,000 users and developers do now that the maximum flexibility ever thought will be in both sides of the VoIP solution?. Even at home (for those who are hobbyist) you could set your own VoIP solution and adapt it to fit your needs.

This is the beginning of a new era of innovative solutions to rise. If you would like to introduce yourself in the world of VoIP, don’t miss some future articles where I’ll be covering the installation details and “obscure” (not really) steps about how to set your own VoIP solution at home.

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following is a sample of the issues resolved in this release:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!

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:

• Arno’s IPTABLES Firewall
• IPv4 and IPv6 routing and filtering
• Traffic Shaping using QoS markings
• VPN Support: IPSec, OpenVPN and PPTP
• Caching DNS Support (dnsmasq)

Supported platforms include:

• Jetway Intel Dual Core Atom boards
• PC Engines alix2d3 / alix2d13
• Soekris Engineering net5501
• VIA C3 and C7 based systems
• Generic x86 PC Hardware

The following telephony hardware cards are supported:

• Digium
• Sangoma
• Rhino Equipment
  

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 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!

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.