Analog Devices launched the new Blackfin BF51x  series, the newest members of their  convergent-processor family.  The Blackfin convergent-processors have superior ability to reduce cost, power consumption, software complexity, and development schedules in a broad spectrum of mobile lifestyle, industrial and instrumentation, portable medical diagnostic and VoIP telephony applications. Blackfins are commonly used in Asterisk- based VoIP appliances like Astfin and Digium AA50.

The newest members of the Blackfin family are the BF512, BF514, BF516 and BF518, they are all  single-core convergent processors that surpass outdated, heterogenous MCU+DSP approaches in reducing part-count, system cost, board space, and power consumption. These new processors are available at clock speeds of up to 400 MHz and include 116 kBytes of RAM plus an optional 4 Mbits of serial (SPI) flash memory. Each also integrates Lockbox™ security for code and content protection.

The BF51x processors feature high clock rates and low power dissipation per unit of processing (MMACs/mW), and like traditional MCUs, these convergent processors are OS and compiler-friendly.

Click here for more details.

Popularity: 4% [?]

Digium and Skype have announced the availability of a new channel driver that will enable Asterisk users to access Skype features integrated with the features and capabilities of Asterisk. The new Skype for Asterisk Beta allows the creation of native Skype trunks inside Asterisk 1.4 or 1.6. The Skype wideband codecs that provide excellent call quality to Skype calls will be built into the module. This will allow the transcoding between Asterisk codecs and the Skype SYLK codec.

Beta testing for the Skype for Asterisk has started and it is limited to a small group selected to represent a broad cross-section of potential users and developers.

The new module requires a per-channel license fee, therefore it is not going to be an open source. The Skype module will be available through the Digium Asterisk Marketplace.

Click here for more information.

Popularity: 2% [?]

Digium has released the AEX410, a modular four-port analog PCI-Express x1 telephony interface card for Asterisk. The AEX410 is the PCI-Express version of Digium’s existing TDM410 product.

The AEX410 provides analog (FXS) ports and trunk (FXO) ports for POTS/PSTN connectivity. It may be connected to the PSTN, to other PBXs, or to analog handset devices with those modules. Optional DSP-based 128ms line echo cancellation for the AEX410 is provided by Digium’s VPMADT032 G.168 module. The optional hardware echo cancellation provides 1024 taps of echo cancellation for superior voice quality.

The naming convention for the AEX410 part numbers is as follows:

AEX4XYZ

Where X indicates the number of FXS (station) modules (ports)
Where Y indicates the number of FXO (trunk) modules (ports)
Where Z indicates either B for bundles not containing DSP-based echo cancellation or E for bundles that do contain DSP-based echo cancellation.

Click here to read the complete announcement

Popularity: 2% [?]

Good news to all Asterisk fans out there, Asterisk Graphical User Interface (GUI)  2.0 has been released and it looks great.

Asterisk, the most popular open source telephony system has been criticized in the past for not having a good GUI and most network administrators and asterisk enthusiasts like me prefer to use the Command Line Interface (CLI) to configure and maintain an asterisk platform.

Click HERE to check the screenshots of the new Asterisk GUI 2.0 

Popularity: 2% [?]

Digium has announced the latest member of the Switchvox appliance family, the Digium|Switchvox AA300 appliance to go along with the AA60 and AA350 models. The AA300 supports up to 150 users and is targeting mid-sized companies. The AA300 model size is 3U half-deph and it can be rackmounted. In addition to 150 users, it supports up to 45 simultaneous calls. It also supports up to 10 recorded calls and up to 15 simultaneous conference calls.

Switchvox AA300 highlighted features:

•  Full coverage by the Digium Exceptional Satisfaction Program, the strongest guarantee in open source telephony, which gives customers their money back if the product is defective or fails to perform as described.
•  VoIP ready—No extra hardware is necessary to connect to SIP or IAX voice over IP trunks.
•  A standard one-year warranty and optional three-year warranty.
•  Support for analog and T1/E1 interfaces gives customers flexibility in working with their existing network.
•  A choice of three subscription plans to suit every company’s support and service needs.

Pricing for the Digium Switchvox AA300 with SMB 3.5 software is $4,240.View the complete announcement here.

Popularity: 1% [?]

How would you like your voice or the voice of the person over the phone to sound like a chipmunk?

Lobstertech Technologies has developed Asterisk Voice Changer which allows you to change the pitch of your voice, or the voice of the called party in realtime, when making phone calls over Asterisk based systems.

The application, VoiceChangeDial, functions as a Dial() replacement except it allows you to change the pitch of your voice. Although the behavior of this application seeks to mimic Dial(), it is not perfect due to the fact that Asterisk does not properly abstract dial and bridging functionality.

Installation Procedure

On your functioning Linux machine running Asterisk with the header files installed to /usr/include/asterisk, run the following commands to install the voice changer as well as its two dependencies:

# install SoundTouch 1.3.1
# you can also install it through your package manager
cd /usr/src
wget http://www.surina.net/soundtouch/soundtouch-1.3.1.tar.gz
tar xvzf soundtouch-1.3.1.tar.gz
cd soundtouch-1.3.1/
./configure –prefix=/usr
make
make install

# install libsoundtouch4c
cd /usr/src
wget http://www.lobstertech.com/code/libsoundtouch4c/releases/libsoundtouch4c-0.4.tar.gz
tar -xzvf libsoundtouch4c-0.4.tar.gz
cd libsoundtouch4c-0.4
./configure –prefix=/usr
make
make install

# install the voice changer
cd /usr/src
wget http://www.lobstertech.com/code/voicechanger/releases/voicechanger-0.6.tar.gz
tar -xzvf voicechanger-0.6.tar.gz
cd voicechanger-0.6
make
make install

# load it in to asterisk
make start

Have fun!

Popularity: 1% [?]