More on Skype for Asterisk

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More on Skype for Asterisk

Continuing the coverage of the big Skype for Asterisk news I covered earlier today... In a nutshell, the Asterisk server acts as a Skype-to-SIP gateway, a very popular requested feature, mapping Asterisk 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. 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 queues. 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.

[section added since Digium's Steve Sokol explained how to handle transfers from IP phones to Skype usernames.]

We've got a couple of ways to do it. The first and most simple way would be to create a local numeric alias for the Skype name. In that case you simply transfer the call to the numeric alias which then sends the call out the Skype channel. The extensions.conf logic looks like this:

exten => 6101,1,Dial(Skype/ssokol.digium)

In the above example the extension number is 6101 and the Skype name to which the call is forwarded is ssokol.digium.

Another mode of transfer would involve a graphical user interface like the Switchvox Switchboard. In that case the user would simply drag and drop the call on an appearance that maps to the Skype name. Under the covers it would use the Manager API to execute the transfer.

I'm sure that there are a number of other modes or techniques that could be used. Our developer community is very good at inventing clever solutions.
[end section added]

When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg 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:

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