A Digium anunciou durante o IP.4.IT, em Las Vegas, Nevada, o Asterisk 1.2. A mais nova versão deste software traz aos seus usuários mais de 3.000 melhorias, atualizações, correções e novas funcionalidades.
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Digium Announces the Launch of Asterisk 1.2
Asterisk 1.2 has over 3,000 improvements, upgrades, fixes, and additions
Digium Inc., the creator of Asterisk® and pioneer of open source telephony, announced Asterisk 1.2, today at the IP.4.IT conference in Las Vegas, Nevada. Asterisk 1.2 is the first major revision to Asterisk since the release of Asterisk 1.0 in September 2004, and includes over 3,000 feature additions and improvements to the overall performance and efficiency of memory usage. Asterisk, the world’s first open source PBX, offers a strategic, highly cost-effective approach to voice and data transport over TDM, IP and other architectures.
"We have been working very hard with the support of the Asterisk community to release version 1.2 of Asterisk," said Mark Spencer, president of Digium and creator of Asterisk. "As Asterisk plays an ever expanding role in the telecommunications industry, it's important to support the rapid development model of open source software – quickly moving features from concept to product while retaining software quality and architectural integrity."
A significant number of changes have been made to the core of Asterisk including code formatting, simplification and documentation. The Asterisk developer community extends all over the world, and the new changes incorporated in Asterisk 1.2 make it easier for new developers to get involved. New features include:
- A number of significant changes to the core to improve performance and memory usage
- Improved voicemail features
- Addition of the DUNDi (Distributed Universal Number Discovery) protocol
- Easier Asterisk configuration
- Creation of a Realtime Database Configuration Storage Engine
- More power added to the Asterisk Dialplan
- Introduction of Asterisk Extension Logic, a new, flexible method for configuring the dialplan
- New interface for dynamic IVR flow control
- Configurable access to general call features
- Improved SIP protocol support
- New features for the IAX (Inter-Asterisk eXchange) protocol
- Use of sound files for native music-on-hold
- Customized CDR Support
- PRI support improvements
Asterisk 1.2 will be available for download from the Asterisk website,FTP and CVS servers after 5:00PM Pacific Standard Time on November 16th.
Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. It also supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure. Using the Inter-Asterisk? eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using packet voice, it is possible to send data such as URL addresses and images in-line with voice traffic, allowing advanced integration of information.
The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.
Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.
Digium solutions reduce the costs of traditional TDM and VoIP implementations through Open Source, standards-based software and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart, and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX™ (Inter-Asterisk Exchange), SIP, MGCP, Cisco Skinny® (SCCP), and H.323 VoIP protocols.