<div dir="ltr">Khomp e Digivoice e Aligera que segure..... asterisk 12 beta liberado... rs</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/9 Asterisk Development Team <span dir="ltr"><<a href="mailto:asteriskteam@digium.com" target="_blank">asteriskteam@digium.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The Asterisk Development Team is pleased to announce the first beta release of<br>
Asterisk 12.0.0. This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases</a><br>
<br>
All interested users of Asterisk are encouraged to participate in the<br>
Asterisk 12 testing process. Please report any issues found to the issue<br>
tracker, <a href="https://issues.asterisk.org/jira" target="_blank">https://issues.asterisk.org/jira</a>. All Asterisk users are invited to<br>
participate in the #asterisk-bugs channel to help communicate issues found to<br>
the Asterisk developers. It is also very useful to see successful test reports.<br>
Please post those to the asterisk-dev mailing list (<a href="http://lists.digium.com" target="_blank">http://lists.digium.com</a>).<br>
<br>
Asterisk 12 is the next major release series of Asterisk. It will be a Standard<br>
release, similar to Asterisk 10. For more information about<br>
support time lines for Asterisk releases, see the Asterisk versions page:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions</a><br>
<br>
For important information regarding upgrading to Asterisk 12, please see the<br>
Asterisk wiki:<br>
<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12</a><br>
<br>
A short list of some of the new major features includes:<br>
<br>
* A new SIP channel driver and accompanying SIP stack named chan_pjsip has been<br>
added. This new channel driver is based on the PJSIP SIP stack by Teluu. It<br>
includes support for the vast majority of features currently in chan_sip,<br>
as well as numerous architectural improvements that alleviate pain points<br>
present in the legacy SIP channel driver. Users who wish to use the new SIP<br>
channel driver are encouraged to read the instructions on installing and<br>
configuring PJSIP for Asterisk on the Asterisk wiki at<br>
<a href="https://wiki.asterisk.org/wiki/x/J4GLAQ" target="_blank">https://wiki.asterisk.org/wiki/x/J4GLAQ</a>. Detailed instructions on configuring<br>
the new SIP stack in Asterisk can be found on the Asterisk wiki as well, at<br>
<a href="https://wiki.asterisk.org/wiki/x/hYCLAQ" target="_blank">https://wiki.asterisk.org/wiki/x/hYCLAQ</a>. Test reports of successful use of<br>
chan_pjsip, with endpoint details, in addition to bug reports, are most<br>
welcome.<br>
<br>
* The Asterisk REST Interface (ARI) has been added. This interface lets<br>
external systems harness the telephony primitives within Asterisk to develop<br>
their own communications applications. Communication with Asterisk is done<br>
through a RESTful interface, while asynchronous events from Asterisk are<br>
encoded in JSON and sent via a WebSocket. More information on ARI can be found<br>
at <a href="https://wiki.asterisk.org/wiki/x/lYBbAQ" target="_blank">https://wiki.asterisk.org/wiki/x/lYBbAQ</a><br>
<br>
* Major standardization of the Asterisk Manager Interface and its events have<br>
occurred within this version. In particular, the names of Asterisk channels<br>
no longer change and are stable throughout the lifetime of the channel.<br>
More information on the changes in AMI can be seen in the AMI 1.4<br>
Specification at <a href="https://wiki.asterisk.org/wiki/x/dAFRAQ" target="_blank">https://wiki.asterisk.org/wiki/x/dAFRAQ</a><br>
<br>
* All bridging within Asterisk is now performed using the Asterisk Bridging API,<br>
which previously was only used by the ConfBridge application. This affords<br>
Asterisk users greater stability, and has resulted in the abstraction of<br>
channel masquerades, renaming, and other internal implementation details. It<br>
also allows for the seamless transition between two-party and multi-party<br>
bridges using core features.<br>
<br>
And much more!<br>
<br>
More information about the new features can be found on the Asterisk wiki:<br>
<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation</a><br>
<br>
A full list of all new features can also be found in the CHANGES file.<br>
<br>
<a href="http://svnview.digium.com/svn/asterisk/branches/12/CHANGES" target="_blank">http://svnview.digium.com/svn/asterisk/branches/12/CHANGES</a><br>
<br>
For a full list of changes in the current release, please see the ChangeLog.<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-beta1" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-beta1</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
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</font></span></blockquote></div><br><br clear="all"><div><br></div>-- <br><span style="font-family:trebuchet ms,sans-serif">Sylvio Jollenbeck<br><font size="1"><a href="http://www.hosannatecnologia.com.br/" target="_blank">www.hosannatecnologia.com.br</a></font></span><br>
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