More on Skype for Asterisk
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[September 25, 2008]

More on Skype for Asterisk

Originally posted on VoIP & Gadgets Blog, here: http://blog.tmcnet.com/blog/tom-keating/asterisk/more-on-skype-for-asterisk.asp.

Continuing the coverage of the big Skype for Asterisk news I covered earlier today... In a nutshell, the Asterisk server acts as a Skype (News - Alert)-to-SIP gateway, a very popular requested feature, mapping Asterisk (News - Alert) SIP-based phones onto the Skype network via the Asterisk Skype channel driver. Technically, you could call Asterisk a Skype-to-IAX gateway as well.

So how does it work?

Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC's Skype software or your IP phone. Similarly, if someone calls your SkypeIn number, both will ring. Further, if someone dials your corporate auto-attendant, and then enters an extension number, it will still ring both your Skype client and your regular IP phone. That's huge! You can be remote and use Skype as your remote IP phone.

Essentially, Skype becomes a softphone extension of the Asterisk IP-PBX (News - Alert). Although, it's important to note that that outbound calls from the Skype client go through the Skype network and not through Asterisk, so it's not a full-fledged softphone application which does inbound & outbound through the same Asterisk IP-PBX - important for call detail records (CDR) that businesses need.

Also, using Skype for Asterisk you can assign Skype IDs/usernames to an Asterisk call queues. So for instance, you can setup 'tmcsupport' or 'tmcsales' Skype usernames and then anyone in the world can call into these call queue. Skype's rich presence will be integrated into Asterisk, but it isn't currently part of the beta, but should be part of the final release. What that would allow is a remote agent to set their presence to Away or Available and then take inbound calls to the Asterisk queue based on their presence.



When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan �berg VP & GM Telecom for Skype said, "The appliances that are out there now have built their solutions on standard Linux client. They've used the public API on that and basically are running many instances of Skype Linux client. Obviously, that's not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that businesses would want them to be. The difference with this solution is that we've built it together to scale and to be reliable."

When asked, "What about video integration?" Danny Wyndam responded, "The beta product that is available today does not support video. It is our plan to be able to support everything you can do in Skype through Asterisk. It's just an evolution of the connector to this platform that we can add the video support."


Danny pointed out that in Asterisk you will be able to define calling rules with least cost routing (LCR) and determine if the call should go out through the T1/PRI/analog trunk or over SkypeOut to save on the costs.

When asked, "How long have you been working on this?", Danny answered that they have been in talks for at least 3 years - but very serious for a few months in integrating Asterisk with Skype.

Here's a shot from Astricon showing it in action:
skype-for-asterisk.jpg

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