Digium Thursday confirmed the release of Asterisk 10, the latest version of the 12-year-old Asterisk open-source telephony platform that's slowly but surely gaining traction in the broader telephony market.
Digium, which is Asterisk's primary developer, announced the release in line with this week's AstriCon conference in Denver. According to Digium, the freely available Asterisk platform has seen millions of downloads in the past few years, including 2 million in 2010 alone.
Asterisk 10, also known as Asterisk version 1.10, is a substantial update focused on the platform's media engine, according to Steve Sokol, Asterisk marketing director at Digium. Developers added a number of new codecs to the platform, including Skype's SILK codec, 32kHz Speex support and pass-through support for CELT, and it can support all types of audio. According to Digium, Asterisk previously operated on 8kHz and 16kHz sampled audio, but can now support 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for audio.
"The new version is actually capable of negotiating the specific details of media," Sokol explained. "That's the biggest thing: a rip-and-replace of the media engine so that it's now capable of doing any kind of media or any video at almost any quality rate."
Also new is a conferencing bridging application called ConfBridge that replaces the previous MeetMe conferencing bridge. ConfBridge supports the full range of codecs and works on any Asterisk 10-based system, regardless of OS, and can be further customized by system administrators and developers. ConfBridge also can relay video of designated speakers to other participants in a conference, provided video-capable SIP devices that use that same codec are in use.
Other new features include amped-up fax capabilities. Asterisk 10 offers T.38 to allow outgoing fax calls from analog fax machines to connect to T.38 fax endpoints over SIP, and incoming T.38 fax calls to be delivered to fax machines. The new Asterisk also adds text message routing to the available text message functions, supporting SIP Message and XMPP protocols.
The new version of Asterisk is now available, and Sokol said he expects the features to make their way into Switchvox, Digium's commercially available, Asterisk-based IP-PBX system, over the next few years. Like previous releases, Asterisk 10 is available for free download and licensed under the GNU General Public License v2.
Digium has seen increasing Asterisk product adoption in line with broader market acceptance of open-source telephony, Sokol said. According to a June 2011 report from market researcher Eastern Management Group, open-source PBXes comprise 18 percent of the overall market. Asterisk-based systems comprise more than 85 percent of those, according to most estimates.
"More and more mainstream businesses are adopting it, whether it's using raw Asterisk or in its several forms, like Switchvox," Sokol said.
The Asterisk community has more on the way, he added, particularly around the Asterisk Scalable Communications Framework (SCF), the new open-source project Digium launched in fall 2010. The goal for SCF is to enable realtime UC applications for enterprises and carriers -- building "an Asterisk solution that is scalable, redundant and fault tolerant," Sokol explained.
Digium also is forging ahead on its Redundancy Series, or R-Series, products, including appliances to enable physical layer failover of telephony connections to Asterisk-based systems. A formal release announcement is coming soon, said Sokol, and MSRP for the R-Series gateway is expected to be about $1,000.