As per usual, Tom Keating gets us more details on the “Skype For Asterisk” beta program I just wrote about… in his update post, Tom explains how it will 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.
His post discusses how you can assign Skype names to Asterisk call queues and then includes this intriguing text:
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.”
If I understand this correctly, this has the potential to be huge! As far as I know, all the existing “Skype-to-PBX” solutions use the rather kludgey solution of basically running multiple instances of the Skype client on the system. Each “Skype trunk” is essentially just a separate instance of the Skype client. As Stefan Öberg indicates, there are serious scaling issues with this approach.
However, this has been the only options developers have had! Skype has not – prior to this (if it works how it sounds like it works) – provided any “back-end API” that would let a system interact directly with the Skype P2P cloud. The only API developers have had is the client API that lets them interact with a local Skype client. So that’s how all the “Skype-to-X” products have been built.
Does this mean that Skype has exposed some additional API that is available through this Skype For Asterisk product? If so, this could be VERY interesting…
Kudos to Tom for providing the updated information.
Technorati Tags:
asterisk, skype, sip, interoperability, voip, telecom
Very nice find. Kudos to Tom. I think the real overall “problem” here is that skype does not break out of particular segments, or end customer types, and has to date, no real “enterprise” successes. If Skype continues to say “make calls”, it may fail to find true breakout. Maybe that’s why the new head guy talks about video?
everything i have read say users will be able to call out with skype credits at ‘regular skypeout rates.’ am i reading right that they are blocking the use of the bundled or so called ‘unlimited’ calling plans? one of the major attractions of using alternate skype trunking solutions has been the ability to take advantage of skypes various ultra inexpensive calling packages. many sip provider offer rates considerably more competitive than skypeout per minute rates.
i am also curious if skype for asterisk has any capability to pass through the caller ID for call that originate on a non skype trunk but are answered on a skype client. i have read that the ability is offered to send calls with various skypenames to help identify them but if the original caller ID can not be passed to the skype client it could deter usage for some in favor of a SIP or IAX client.
PING:
TITLE: Skype Kills Off “Skype For Asterisk” – A Sign of the New Microsoft Era?
BLOG NAME: Disruptive Telephony
Word breaking out right now from multiple sources is that Skype has killed off the Skype for Asterisk product developed in conjunction with Digium. In an email sent by Digium product management that was subsequently posted on web sites (including…
PING:
TITLE: Clarifying how Asterisk could possibly be used as a Skype-to-SIP gateway
BLOG NAME: Disruptive Telephony
After my post yesterday about “Skype for Asterisk” (and the update post) and the potential it allows for SIP interoperability via Asterisk, I’ve received a few comments that seemed to interpret what I wrote as somehow indicating that the Skype