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Digium Announces Asterisk 13 Open Source Communications Software

Digium Announces Asterisk 13 Open Source Communications Software

HUNTSVILLE, AL, October 22, 2014—Digium®, Inc., the Asterisk® Company, at its annual AstriCon users and developers conference, announced the release of Asterisk 13, a production-ready Long Term Support (LTS) release that builds upon the significant changes made in the previous development release, Asterisk 12. Refined over the last year, Asterisk 13 is a stable, feature-rich, and developer-friendly platform for adding media capabilities to any business application. Asterisk 13 provides hundreds of new features and improvements made since the previous LTS release including the new and improved Asterisk REST Interface (ARI), a re-architected bridging and media core, remote administration enhancements, and numerous improvements to its PJSIP-based SIP channel driver.

“Asterisk 13 represents the most ambitious release of Asterisk yet,” said Matt Jordan, project lead for Asterisk. “This release is the culmination of the efforts of thousands of developers and users worldwide. All of them played an integral role in making Asterisk 13 a reality. We are very proud of what’s been accomplished in this next, great release of Asterisk.”

“When we set out to build Asterisk 13, our goal was to polish all of the massive improvements previously made in Asterisk 12 into something that can be relied on by anyone who needs to develop, deploy, or integrate media into business applications,” said David Deaton, vice president of engineering for Digium.

Asterisk 13 is the first LTS release since Asterisk 11 in 2012. Even though it is substantially improved in terms of architecture and features, users migrating to Asterisk 13 can enjoy all the same capabilities available in previous versions of Asterisk. Developers can upgrade to Asterisk 13 and keep their existing functionality while exploring the new engine and building custom applications using new interfaces.

First introduced in Asterisk 12, ARI has been extended and enhanced in Asterisk 13 to create a fully-featured, production-ready API for developing applications that use Asterisk as a media engine. Developers can now use ARI to build their own custom communications applications such as call queues, voicemail, conferencing and more.

Asterisk 13 includes an improved bridging and media core that greatly simplifies call tracking for developers using Channel Event Logging (CEL) and Asterisk Manager Interface (AMI) capabilities, each of which have been improved. For users, the bridging core of Asterisk 13 features greatly-improved transfer handling, including the ability to perform attended-transfers into multiple parties.

Other new features in Asterisk 13 include the conveyance of security events over AMI, allowing systems to monitor the security state of Asterisk in real time. Resource List Subscription (RLS) support has been added to the PJSIP stack, decreasing the load on Asterisk systems and improving performance. Further, the PJSIP stack has been improved to allow clustering of shared device state and message waiting indicators between multiple Asterisk systems.

Asterisk 13 is currently available for download from the Asterisk web site, www.Asterisk.org. For more information, documentation and usage samples, as well as a complete list of new features, changes and upgrade notes, visit  https://wiki.Asterisk.org/wiki/display/AST/Asterisk+13+Documentation.
 

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