For Immediate Release
Digium Unveils the Latest Version of Asterisk
Asterisk 1.4 features improved performance, scalability and interoperability
HUNTSVILLE, AL and BOSTON, MA — (September 12, 2006) — Digium Inc., the Asterisk company, today unveiled Asterisk 1.4, at the Fall VON conference in Boston, Massachusetts. Originally created by Digium as the industry’s first open source telephony platform, Asterisk offers a more flexible and cost-effective alternative to traditional voice and data solutions.
Asterisk 1.4 is the first major release of Asterisk since the release of Asterisk 1.2 in November 2005. With over 20 new functionality additions including IPFAX compatibility, unified messaging capabilities and Jabber/Jingle/GoogleTalk protocol compatibilities, Asterisk 1.4 features overall quality and performance improvements, as well as increased scalability and interoperability.
“This is by far the best version of Asterisk to date,” said Mark Spencer, president of Digium and creator of Asterisk. “With the support of the Asterisk community, we have been able to develop an advanced platform that will make it even easier for users to migrate to VoIP, especially those in the enterprise community.”
Specific enhancements featured in Asterisk 1.4 include:
*Generic Jitter Buffer- improves the quality of a call during network congestion.
*Asterisk Extension Language Version 2- simplifies programming and dial plan configuration.
*T.38- allows IP FAXes to pass through the server.
*Jabber/Jingle/GoogleTalk- supports compatibility with all of these networks.
*Increased language capabilities- offers new language capabilities in English, Spanish and French as well as new sounds and improved sentence structure support.
*Unified Messaging- integrates voicemail, email, and fax into a central mailbox where users can send, retrieve and manage all of their messages using any communication device.
*Whisper Paging- allows for selective, pre-programmed call interruption with controlled volume levels and muting capabilities.
Additionally, Asterisk 1.4 now includes variable length DTMF support (touch-tone signaling for IVR applications), the option for programming shared line appearance, centralized RADIUS storage for call detail records, a built-in web manager interface and a simplified, single user configuration for SOHO/SMB users. Asterisk 1.4 also offers increased memory usage and performance improvements such as improved interoperability of SIP call transfers, IAX2 scalability improvements, enhanced IAX2 media stream capabilities (enabling direct audio communication between IAX devices while eliminating server involvement and maintaining billing and control functionalities), Cisco SCCP support, SNMP monitoring, and RTP native bridging capabilities.
Support and Availability
Asterisk 1.4 will be available for download on Digium’s website (www.digium.com) in October. Digium will also be demonstrating Asterisk 1.4 in Booth #819 at the Fall VON conference.
About Digium
Digium, the Asterisk company, is the original creator and primary developer of Asterisk, the industry’s first open source telephony platform. Digium provides hardware and software products, including the Asterisk Business Edition, its professional grade version of Asterisk, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services including consulting, technical support and custom software development services.
Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium’s offerings include legacy PBX, IVR, auto attendant, next generation gateways, media servers and application servers. Additional information can be found at www.digium.com.
About Asterisk
Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed to from open source software engineers around the world. Currently boasting over 1 million users, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.
The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.
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Media Contacts:
Julie Webb
Digium, Inc.
256-428-6203
jwebb@digium.com
Janine Savarese
MRB Public Relations
732-758-1100 Ext. 103
jsavarese@mrb-pr.com
