[AsteriskBrasil] Digium releases Asterisk 1.2

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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.. A number of significant changes to the core to improve 
performance and memory usage
        b.. Improved voicemail features
        c.. Addition of the DUNDi (Distributed Universal Number Discovery) 
protocol
        d.. Easier Asterisk configuration
        e.. Creation of a Realtime Database Configuration Storage Engine
        f.. More power added to the Asterisk Dialplan
        g.. Introduction of Asterisk Extension Logic, a new, flexible method 
for configuring the dialplan
        h.. New interface for dynamic IVR flow control
        i.. Configurable access to general call features
        j.. Improved SIP protocol support
        k.. New features for the IAX (Inter-Asterisk eXchange) protocol
        l.. Use of sound files for native music-on-hold
        m.. Customized CDR Support
        n.. PRI support improvements
      Availability
      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.

      About Asterisk
      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 (IAXT) 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.

      About Digium
      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 IAXT (Inter-Asterisk Exchange), SIP, MGCP, Cisco Skinny® (SCCP), 
and H.323 VoIP protocols.


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