- About VoIP
- What is VoIP and what it can do for you
- Introduction to VoIP (video)
- Why should you switch to VoIP services?
- Analog Telephony
- Digital Telephony
- What is SIP?
- How to start with VoIP telephony
- Web based VoIP
- How to choose a right VoIP provider?
- Wi-Fi network and VoIP
- VoIP Codecs
- Free sip account
- Confidential calls
- VPN: UDP or TCP?
- Mobile VoIP
- VoIP on your mobile
- Asterisk IP-PBX
- Who we are?
- How to start
- Free SIP account
- Configs
Asterisk 10.0.0-rc1 Now Available!
Digium releases Asterisk 10.
As you may already know Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other communications systems. It has been downloaded millions of times, including two million last year alone, establishing Asterisk as the most popular open source telephony engine.
The most important new feature in Asterisk 10 is its wide-band media engine. Digium has replaced Asterisk’s telephony-grade media engine with a more advanced one, providing support for studio-quality audio and a nearly unlimited number of codecs. By supporting high and ultra high-definition voice, Asterisk can now be used to power communications applications that would have otherwise required specialized or expensive equipment and service in order to convey nuances in speech or emotion. Digium has also updated Asterisk’s media support for Asterisk 10, with several new codecs, including Skype’s SILK codec, 32kHz Speex support and pass-through support for CELT.
Built with open source community support
Digium is advancing Asterisk with version 10, while simultaneously leading work on the Asterisk Scalable Communications Framework. Asterisk SCF will allow developers to create real-time communications applications that include voice, video and text that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions.
Asterisk 10 makes its debut at AstriCon, the Asterisk User Conference & Expo, in Denver. Hundreds of attendees, including software and PBX developers, enterprise IT pros, systems integrators and call center and CRM developers, welcomed the announcement. In its eighth year, AstriCon is offering conference tracks focusing on technical information, carriers and call centers, cloud computing, commerce, government, enterprise and the Asterisk ecosystem. Developer conferences geared toward contributors to the Asterisk and Asterisk SCF projects are also taking place during this year’s AstriCon.
Asterisk 10 is available for free download and is licensed under the GNU General Public License v2.
New features in Asterisk 10
Asterisk 10 offers developers, integrators, resellers and telephony pros a range of new capabilities. A few include:
- New media engine—Asterisk 10 supports more media types and virtually any type of audio. The overhaul to the media engine allows Asterisk to support a nearly unlimited number of codecs.
- More codecs—The platform includes new codecs, including the wideband version of Speex, Skype’s super-wideband SILK and pass-through support for several CELT variants.
- Additional sampling rates—Asterisk previously operated on 8 and 16 kHz sampled audio, but now supports super- and ultra-wideband sampling rates as file format types for file playback or recording. Asterisk now supports 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for superb audio quality.
- New conferencing application—Digium replaced the MeetMe conferencing bridge with an HD-capable intelligent bridge application called ConfBridge. It supports all codecs and conference rates and works on any Asterisk 10 system, regardless of operating system or architecture. Intelligent mixing algorithms provide each participant with the optimal audio quality for their connection. Also, ConfBridge is fully customizable, so systems administrators and integrators can configure call-in menus on a caller-by-caller basis.
- Support for videoconferencing—ConfBridge relays video of a designated speaker or the current speaker to other participants in the conference. Video-capable SIP devices that use the same codec are required.
- Significant new fax capabilities—Asterisk 10 includes T.38 gateway capabilities that allow outgoing fax calls from analog fax machines to be connected to T.38 fax endpoints over SIP and incoming T.38 fax calls to be delivered directly to fax machines. This allows for more straightforward integration of fax capabilities into an Asterisk system and allows users to get delivery confirmation from other fax machines.
- Text message routing—Asterisk has long been able to send and receive text messages, but can now route messages as well. Asterisk 10 supports the SIP MESSAGE and XMPP protocols, allowing it to act as a text messaging server and bridge between different messaging protocols.
Source: Originally Submitted by asteriskteam on Thu, 11/10/2011
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 10.0.0. This release candidate is available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk/
All Asterisk users are encouraged to participate in the Asterisk 10 testing
process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/jira. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
A short list of features includes:
- T.38 gateway functionality has been added to res_fax.
- Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far. - New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
(More information available at
https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) - Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application. - Support for defining hints has been added to pbx_lua.
- Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
- Much, much more!
A full list of new features can be found in the CHANGES file.
http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
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