Waiting until the last day, Digium and Skype have announced a new channel driver that will soon go into beta testing. The new Skype for Asterisk will allow you to create a native Skype trunk within Asterisk 1.4 or 1.6. This new module will not be open source as it will require a per-channel license fee (pricing not yet set). Beta testing for a select group of users will begin shortly. The Skype module will be available through the Digium Asterisk Marketplace. The Skype channel will allow you to not only use Skype-In and Skype-Out minutes but will also allow you to make and receive free calls from Skype clients and other Skype channel users. Skype’s wideband codecs, that provide for excellent call quality that has helped make Skype successful will be built into the module to allow for transcoding between Asterisk codecs and the Skype SYLK codec. You will also be able to setup multiple Skype names on the channel to provide for routing within the Asterisk PBX based on the Skype handle that was called.

For more information, please visit http://astricon.net/skype