Digium betas Skype/Asterisk integration

Digium, developer of Asterisk, the open source PBX, has announced a partnership with Skype and a beta version of Skype For Asterisk. The gateway, announced at Astricon 2008 is available in an invitation-only beta programme; Asterisk users can apply to join the beta on the Astricon web site.

According to a report on Skype Journal, the gateway will allow a rich integration of Skype and Asterisk features; users will be able to create Skype ids that can be used as "global 800 numbers" with incoming calls being handed off to call centres, voice mail or conferences by Asterisk. Skype will also be an option for Asterisk's least-cost call routing, so outgoing calls can be routed via Skype where it is cheapest. Where calls are made from one Skype client to another, the gateway will automatically use Skype's HD audio mode allowing for clearer calls.

Details on the pricing of the gateway are unavailable at this time, though Skype will supply "Premium Packages" tailored for specific markets with enhanced administrative controls. Third parties have already had Skype gateway connections available, including ChanSkype. It is also unclear what the licensing terms of the Skype for Asterisk module will be. Again, applications already exist under open source licenses to allow for SIP/VOIP connections to be made to Skype such as SipTheeSkype.

(djwm)