Skype for Asterisk Launches

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Randy Savicky
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Skype for Asterisk Launches

Skype and Digium have hooked up to bring Skype to Asterisk called Skype For Asterisk. Skype For Asterisk launched minutes ago enables Asterisk users to get access to Skype features coupled with the capabilities of Asterisk. For example, the beta version of Skype For Asterisk will allow customers to make, receive and transfer Skype calls from within Asterisk systems using their existing hardware; enable inbound calling solutions like free click-to-call from company websites or virtual offices; and manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Hey, I guess I was right in my (Astricon) prognostications earlier today about it having to do with Skype.

The Skype For Asterisk Beta program begins today. Asterisk users, system administrators and developers are invited to apply to participate at http://www.astricon.net/skype

I'm trying to figure out how you transfer a call to a Skype username (i.e. tkeating) using a traditional (Asterisk) IP phone with no keyboard - just a numeric keypad. Of course, maybe the transfer feature is only to other Asterisk extensions or outside phone numbers and you can't initiate calls to Skype usernames. Of course, I'm guessing that you can map inbound Skype calls to usernames to specific Asterisk IP phone extensions.

[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]

Update (1pm): Some other thoughts...
Will Skype for Asterisk work exclusively on Digium's flavors of Asterisk (AsteriskNOW, Switchvox, etc.) or will it also work on trixbox CE, PBX in a Flash, etc? Is the Skype channel driver licensed by Digium or is it a free driver, which can then be used on other Asterisk distros. Since Asterisk offers a free version of their open source solution, I'm going to have to assume the Skype channel driver will also be free.

Update (1:20pm): Some info from TMCnet reporters at Astricon
  • Majority of questions were about access to code. Mark says their will be some limited access.
  • Caller ID - they say it can work.
  • Number portability - Oberg says that is a 'local issue' and not built in to this beta.
  • No pricing announced.
  • Commercial license model, Not open source.

Update (2:58pm) Additional info from an interview with Skype & Digium:
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.

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.

News release after the jump...

Digium®, creator and primary developer of Asterisk®, the leading open source telephony platform, and Skype™, the leading global Internet communications company, today announced the beta version of Skype For Asterisk, which will allow the integration of Skype functionality into Digium's Asterisk software and enable customers to make, receive and transfer Skype calls from within their Asterisk phone systems.

"Throughout our individual histories, Skype and Asterisk have each disrupted conventional communication methods through innovative, cost-effective solutions," said Stefan Öberg, vice president and general manager for Skype Telecom and Skype for Business. "We are excited to be working together with Digium to offer small and mid-sized businesses an even more powerful communications solution to conduct business worldwide."

Specifically, the beta version of Skype For Asterisk is an add-on channel driver module that integrates Skype Internet calling with Asterisk-based telephony products. Skype For Asterisk also complements small and mid-sized business users' existing services by providing low rates for calling landline and mobile phones around the world.

"Working together with Skype, our goal is to help businesses boost productivity and reap the rewards of feature-rich telephony software, all while saving a substantial amount of money," said Danny Windham, CEO of Digium, the creator and sponsor of Asterisk. "The Skype For Asterisk beta program is a first step towards adding Skype capabilities to Asterisk-based phone systems and enabling them to reach more than 338 million Skype users."

The beta version of Skype For Asterisk will enable business users to:
• Make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware.
• Complement existing services with low Skype global rates (as low as 2.1US¢ per minute to more than 35 countries worldwide).
• Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN through Skype's online numbers.
• Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Following the beta period when the product is released, Skype For Asterisk will be sold and distributed by Digium and its worldwide network of resellers.

Live at AstriCon
Stefan Öberg will provide the first public demonstration of Skype For Asterisk during his keynote address today at AstriCon, the annual Asterisk user and developer conference. AstriCon attendees are also invited to stop in and see a demonstration of Skype For Asterisk at the Skype booth on the expo floor.

Skype For Asterisk Beta Program
The Skype For Asterisk beta program begins today; Asterisk users, system administrators and developers are invited to apply at http://www.astricon.net/skype. The initial beta is limited to a select number of users, developers and integrators.


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