Two announcements in the news today from Digium: the company has expanded its cloud offering to include IP trunks and, in separate news, announced the release of Asterisk 13 with hundreds of new features.
Digium SIP Trunking, now offered as part of the latest Digium Cloud Services (DCS), provides VoIP connectivity for Switchvox, Digium’s Unified Communications (UC) system. Digium SIP Trunking is available in two pricing plans. The channelized plan offers unlimited usage at a flat rate. A metered plan is also available for customers that need more flexibility. Digium SIP Trunking also provides unlimited free calls between DCS customers. Pricing starts at $18.45 per month for channel plans with unlimited minutes, and $0.015 per minute for metered plans.
The SIP trunking features include nationwide DIDs, 911-enabled DIDs, and toll-free DIDs; local and toll-free numbers can be ported from other carriers. Multiple codecs and T.38 fax pass-through are also supported. Digium SIP Trunking is available in the U.S. lower 48 states on October 22, 2014.
Digium has also announced Asterisk 13 as a Long Term Support (LTS) release. Asterisk 13 improvements include 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. 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.
Commenting in a statement on the latest release, Matt Jordan, project lead for Asterisk, said “Asterisk 13 represents the most ambitious release of Asterisk yet. 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.”
Asterisk 13 is currently available for download from the Asterisk web site, www.asterisk.org.