Month: January 2011

Recently, Digium has announced both the release of Asterisk 1.8 and the development of Asterisk SCF, billed as the world’s first high performance, distributed, scalable, fault-tolerant, open source communications framework. Recently we spoke with Steve Sokol, Digium’s Marketing Director for Asterisk and Custom Telephony Solutions to learn a little more about what we can expect from this software and the project going forward. Thanks for joining us today Steve. So first why don’t you tell VoIPon’s listeners, what is new in Asterisk 1.8? Well we’ve got kind of a laundry list of new features we’ve added into Asterisk 1.8. I suppose at the very top of that list I would put in full secure calling with secure RTP, so that is the biggest thing. Another thing that is really important is IPv6. There aren’t a whole lot of phones out there yet that support it, but there is a huge push to try to get IPv6 enabled for telephony systems: phones, call control mechanisms, like Asterisk, etc., basically because the world is running out of IPv4 addresses. That is pretty critical. We’ve added a number of ISDN related features, and we’ve actually applied all of those to SIP as well. So anything that you can do on an ISDN network, you should be able to do on a SIP network also. One of them is call completion services. This...