Asterisk software is the world�s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free.
Asterisk is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP.
Asterisk was created by Mark Spencer of Digium, Inc in 1999. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.
Asterisk as a switch (PBX)
Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.
Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk as a gateway
It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk's modular architecture allows it to convert between a wide range of communications protocols and media codecs.
Asterisk as a feature/media server
Need an IVR? Asterisk's got you covered. How about a conference bridge? Yep. It's in there. What about an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok.
Asterisk in the call center
Asterisk has been adopted by call centers around the world based on its flexibility. Call center and contact center developers have built complete ACD systems based on Asterisk. Asterisk has also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more.
Asterisk in the network
Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECS) and even first-tier incumbents have discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk have helped reduce costs and enabled flexibility.
Asterisk everywhere
Asterisk has become the basis for thousands of communications solutions. If you need to communicate, Asterisk is your answer.
What's New in This Release: [ read full changelog ]
· The Asterisk Development Team is proud to announce release of Asterisk 1.4.24, and is available for immediate download at http://downloads.digium.com/
· In addition to other bug fixes, this release candidate fixes several crash issues, and resolved some remaining issues related to call pickup and call parking that were discovered after the release of Asterisk 1.4.23. In addition, issues related to chan_iax2, and regressions introduced to the 'h' extension have been resolved.
· This release marks the first inclusion of the release summary files which will be included in all future releases. The purpose is to give a clearer overview of the changes that have taken place between the current and previous release, which issues have been closed, and which community members were involved with issue submission, code commits, and issue testing. Additionally, a diffstat at the end of the file shows at a brief glance the number of changes made to files between the previous and current releases.
· For a summary of the changes in this release, please see the release summary. For a full list of changes in this release, please see the ChangeLog.
· The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help!
· Paging application crashes asterisk. Closes issue #14308. Submitted by bluefox. Tested by kc0bvu. Patched by seanbright.
· Crash in VoiceMailMain if hangup occurs before a valid mailbox number is entered (IMAP only). Closes issue #14473. Submitted by, and patch provided by dwpaul.
· Incoming Gtalk calls fail. Closes issue #13984. Submitted by, tested, and patched by jcovert.
· Realtime peers are never qualified after 'sip reload'. Closes issue #14196. Submitted by, tested, and patched by pdf.
· SIP Attended Transfer fails. Closes issue 14611. Submitted by, tested, and patched by klaus3000.
