Well, I am pleased to announce that Skype for Asterisk has production released! It has been a long road of field testing under many use cases but all of us at Digium believe the wait for the general public was worth it. The development effort in conjunction with Skype has produced a stable and feature rich product that has the best call quality of any solution in the market.
For those of you who aren’t familiar with Skype for Asterisk (SfA), it is an add-on channel driver that integrates Skype calling with Asterisk-based telephony systems and allows businesses to build a presence in the Skype community. Once a business is connected to the Skype community via SfA, free calls to the 400 million + Skype users and low rates for calling landlines and mobile phones are available to lower their telecommunications costs.
The SfA product will be the only solution that integrates Asterisk directly with Skype with no external gateway hardware. This is not a “proxy” solution and the call quality will be superior to anything else on the market. Customers will have the ability to make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware and existing Asterisk configurations: Skype calls become just another Asterisk call.
Some of the features that are supported in this release are:
- Make Skype-to-Skype calls.
- Receive calls with online numbers (SkypeIn).
- Make world-wide PSTN calls to landline and mobile phones (SkypeOut).
- Make and receive multiple concurrent Skype calls from the same Skype account.
- DTMF support for incoming and outgoing calls.
- Read Skype profile fields from incoming calls.
- Set and retrieve online status.
- Set privacy settings.
- Handle incoming Skype calls using Asterisk applications such as voicemail, ACD, MeetMe conferencing, etc.
- Simultaneous access from both Asterisk and the Skype desktop client.
- Trunk calls between Asterisk servers over Skype.
- Supports G.711 and G.729 (included) codecs.
During the extended beta testing period, there were an abundance of users and test case scenarios applied. Users from all parts of the world participated. One of our favorite uses of SfA involved a solution that implemented both fixed to mobile convergence and dynamic contact routing through Asterisk. The Asterisk-based PBX was integrated into a customer’s traditional telephony service, SIP extensions, Skype users and mobile network. A custom script was developed that attempted to find the user where they are available whether on their SIP client (via their desk-phone or mobile), their Skype account (on their PC or mobile), direct to their mobile or to their voicemail (which is sent to their e-mail account). The order of this routing is based on individual preference and cost efficiency to the business.
So what’s next? Digium and Skype are continuing to collaborate by listening to our customer’s suggestions to enhance SfA for the next release so it can continue to solve problems for businesses. We are all excited to bring this functionality to Asterisk servers everywhere and will continue to enhance the features.