Mashups are fun to work on these days. We rely heavily on both Twitter and Sykpe on a daily basis. Sometimes tweets need to escalate to a phone call.
Rather than forwarding someone a tweet with your Skype username and contact instructions, we decided to hack together a simple Twitter/Skype mashup called http://twype.me. This mashup allows you to link your skype username to your twitter username and when clicked, Skype opens and begins dialing.
How do you use Twype.Me? It’s simple! Go to http://Twype.Me and sign into Twitter. (We use the new OAuth feature so that you do not need to give us your password!) Once signed up on Twype.Me you simple enter your Skype username. That’s it - now you are ready to begin sharing the Twype.Me/twitter_username to initiate calls directly from your tweets or emails or webpages!
You can call my Twitter account via http://twype.me/chrismatthieu
Chris Matthieu had a chance to attend Astricon, Digium’s annual Asterisk open source conference. Besides meeting a bunch of geeky-cool telephony engineers, he was able to take-in a dozen or so presentations. Skype even made a major announcement at the conference - a Skype / Asterisk partnership. Now Asterisk users can call Skype users directly from the switch!
The future of telephony is very bright. Whether Asterisk will the dominant killer application in this space is yet to be determined but it’s gaining speed and adoption. New open source switch projects making headway include: FreeSwitch and Yates. These solutions appear to handle 10x the call volumes per server and include very open APIs.
One of the interesting questions that buzzed around the conference was whether or not we would soon see a Telephony 2.0 phase much like Web 2.0 has been experiencing for several years now. Our answer to this question is definately YES!
Stay tuned…
Have you heard about FreeSwitch? It’s an open source Asterisk competitor that is positioned as more of a soft switch rather than a PBX. It appears to have a more scalable design and perhaps even 10x more call processing power per server. I can’t find any information on their site where FreeSwitch supports analog or digital telephony boards. At first I thought that this would be a problem for the platform but after giving it more thought I think that this is perfect in this Web 2.0 age. Everything is moving to the network as the platform and SIP is the VoIP protocol of choice. Could we create a PhoneSystem.net solution running on FreeSwitch?
Let me know (chris [at] getvocal.com) if you are interested in helping with this endeavor!
