Just last week, Digium announced a partnership with Skype yielding the new "Skype for Asterisk" beta. This exciting announcement came from the annual Digium Asterisk conference, Astricon 2008, in Phoenix Arizona.
I had the recent opportunity to talk with Digium CEO, Danny Windham about the new announcement. Through the conversation, I was able to obtain and analyze some of the new details that have evolved since the original announcement. Danny clearly illustrated that this new integration yields many innovative uses and features that will not only benefit users from a functional standpoint, but from a ROI perspective as well.
In the last few months, I've been personally analyzing Skype's track toward the business market. Back in June, I wrote an article titled "Why Skype Doesn't Scale". (http://www.nwwsubscribe.com/community/node/28313) The new Skype for Asterisk initiative simply resolves nearly 100% of the business scalability and integration issues that were previously questioned.
Let's revisit some of the key features and functionality that came from the original press release:
* 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 2.1USĀ¢ 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.
Mr. Windham re-iterated the significant cost savings of the Asterisk platform in the recent economic tough times. Now, with further integration such as "Skype for Asterisk", there is even more of a push towards open-source-based solutions to increase feature sets and reduce overall telecommunication costs. Danny recently authored a blog post on this subject on the Digium site:
http://blogs.digium.com/2008/10/01/economic-meltdown-%E2%80%93-friend-or-foe-of-open-source/
So, what are the technical and implementation details, you ask? Technically, the Skype for Asterisk application integrates as a channel driver, similar to SIP or IAX channel drivers. Call control and other functionality is typically handled from the Asterisk dialplan.
This project is still in beta, however, a fast turnaround to production is expected. Digium is presently looking to recruit beta testers from diverse technical environments to load-test the product, readying it for the mainstream release.
Bottom-line: this partnership and innovation is another nail in the traditional IP-PBX coffin. Users demand interoperability between platforms of various technology and implementation. This is becoming yet another definite reality with the Skype for Asterisk project.
Nickasch has been very involved in IT since he was just 13. His current and previous consulting experience includes systems architecture, virtualization, and converged networks for the financial, education, and healthcare industries. Matthew currently attends the University of Wisconsin-Platteville, where he also works as a network management assistant. While his interests include directory services and routing protocols, Nickasch's focus is on converged networks and voice over IP.
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