<div dir="ltr">para os curiosos...! rs<div><br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Asterisk Development Team</b> <span dir="ltr"><<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>></span><br>
Date: 2013/11/25<br>Subject: [asterisk-dev] Asterisk 12.0.0-beta2 Now Available!<br>To: <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br><br><br>The Asterisk Development Team is pleased to announce the second 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>
As Asterisk 12 has a more flexible feature policy, a number of new features<br>
have been included in the second beta release that were not present in<br>
the first beta release. These include:<br>
<br>
* Numerous new features and improvements to ARI, including the ability to<br>
manipulate and query device state, snoop on channels (similar to the ChanSpy<br>
dialplan application), and improvements to the overall consistency of the<br>
RESTful API.<br>
<br>
* New AMI commands for the PJSIP stack, including:<br>
- PJSIPShowEndpoint/PJSIPShowEndpoints for querying endpoint information<br>
- PJSIPShowRegistrationsInbound/PJSIPShowRegistrationsOutbound for querying<br>
registration information<br>
- PJSIPShowSubscriptionsInbound/PJSIPShowSubscriptionsOutbound for querying<br>
subscription information.<br>
<br>
* Major improvements to sip_to_pjsip.py, a sip.conf to pjsip.conf conversion<br>
script. Users migrating to the PJSIP stack can use this script to aid in the<br>
migration process from an existing sip.conf configuration file.<br>
<br>
* Standardization of the PJSIP configuration parameters. The configuration<br>
option schema for pjsip.conf now uses snake case for all of its options. This<br>
will impact current beta testers, as options such as 'fromuser' will now be<br>
'from_user', etc. Please verify your options against the documentation on the<br>
Asterisk wiki at: <a href="https://wiki.asterisk.org/wiki/x/ZwWUAQ" target="_blank">https://wiki.asterisk.org/wiki/x/ZwWUAQ</a><br>
<br>
* Many bug fixes reported by the Asterisk community. Thank you for your<br>
collaboration and participation in testing the first Asterisk 12 beta release!<br>
<br>
For more information on the Asterisk 12 release policy, please see the Asterisk<br>
wiki:<br>
<br>
<a href="https://wiki.asterisk.org/wiki/x/1YRHAQ" target="_blank">https://wiki.asterisk.org/wiki/x/1YRHAQ</a><br>
<br>
In addition to these improvements in the second beta release, Asterisk 12<br>
contains many new major freatures. A short list of some of these features<br>
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-beta2" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-beta2</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<span class="HOEnZb"><font color="#888888"><br>
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