While we wait for Digium's official SIP-to-Skype gateway, Nerd Vittles today informed me about his very cool recipe that you can use today to build your own free SIP-to-Skype gateway enabling you to use your SIP-based desktop phones connected to Asterisk to make Skype inbound/outbound calls.

Part of the recipe uses SipToSis - SIP to Skype Gateway Bridge Proxy. SipToSis is a piece of software which Nerd Vittles points out "forms the lynchpin of Gizmo's offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make."

Nerd Vittles explains in his tutorial:

When we're finished, you'll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You'll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server.

In summary, with this tutorial you'll be able to dial Skype users, as well as receive incoming calls from any Skype user! Nerd Vittles' recipe should work on just about any Asterisk-based system. I might have to try this recipe myself later on today. Good stuff!