[asterisk: 2/2] Merge remote-tracking branch 'origin/master'

Wed Sep 26 16:35:32 UTC 2012

commit 8c66d94a79850d417bd4c354195c4ca02e5ed9a0
Merge: 4dddb6f d0b2d25
Author: Jeffrey C. Ollie <jeff at ocjtech.us>
Date:   Wed Sep 26 11:35:09 2012 -0500

    Merge remote-tracking branch 'origin/master'
    
    Conflicts:
    	asterisk.spec

 asterisk.spec |   11 +++++++++--
 1 files changed, 9 insertions(+), 2 deletions(-)
---
diff --cc asterisk.spec
index ddd2b40,d3af846..001342d

--- a/asterisk.spec
+++ b/asterisk.spec
@@@ -31,7 -31,7 +31,7 @@@
  Summary: The Open Source PBX
  Name: asterisk
  Version: 11.0.0
--Release: 0.3%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
++Release: 0.4%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
  License: GPLv2
  Group: Applications/Internet
  URL: http://www.asterisk.org/
@@@ -1379,96 -1378,9 +1383,99 @@@ f
  %{_libdir}/asterisk/modules/app_voicemail_plain.so
  
  %changelog
- * Wed Sep 26 2012 Jeffrey Ollie <jeff at ocjtech.us> - 11.0.0-0.3
++* Wed Sep 26 2012 Jeffrey Ollie <jeff at ocjtech.us> - 11.0.0-0.4
 +- The Asterisk Development Team is pleased to announce the second beta release of
 +- Asterisk 11.0.0.  This release is available for immediate download at
 +- http://downloads.asterisk.org/pub/telephony/asterisk/releases
 +-
 +- All interested users of Asterisk are encouraged to participate in the
 +- Asterisk 11 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 11 is the next major release series of Asterisk.  It will be a Long
 +- Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
 +- support time lines for Asterisk releases, see the Asterisk versions page:
 +- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
 +-
 +- For important information regarding upgrading to Asterisk 11, please see the
 +- Asterisk wiki:
 +-
 +- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
 +-
 +- A short list of new features includes:
 +-
 +- * A new channel driver named chan_motif has been added which provides support
 +-   for Google Talk and Jingle in a single channel driver.  This new channel
 +-   driver includes support for both audio and video, RFC2833 DTMF, all codecs
 +-   supported by Asterisk, hold, unhold, and ringing notification. It is also
 +-   compliant with the current Jingle specification, current Google Jingle
 +-   specification, and the original Google Talk protocol.
 +-
 +- * Support for the WebSocket transport for chan_sip.
 +-
 +- * SIP peers can now be configured to support negotiation of ICE candidates.
 +-
 +- * The app_page application now no longer depends on DAHDI or app_meetme. It
 +-   has been re-architected to use app_confbridge internally.
 +-
 +- * Hangup handlers can be attached to channels using the CHANNEL() function.
 +-   Hangup handlers will run when the channel is hung up similar to the h
 +-   extension; however, unlike an h extension, a hangup handler is associated with
 +-   the actual channel and will execute anytime that channel is hung up,
 +-   regardless of where it is in the dialplan.
 +-
 +- * Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
 +-   allows you to execute a dialplan subroutine on a channel before a call is
 +-   placed but after the application performing a dial action is invoked. This
 +-   means that the handlers are executed after the creation of the callee
 +-   channels, but before any actions have been taken to actually dial the callee
 +-   channels.
 +-
 +- * Log messages can now be easily associated with a certain call by looking at
 +-   a new unique identifier, "Call Id".  Call ids are attached to log messages for
 +-   just about any case where it can be determined that the message is related
 +-   to a particular call.
 +-
 +- * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
 +-   Asterisk. Unlike traditional ACLs defined in specific module configuration
 +-   files, Named ACLs can be shared across multiple modules.
 +-
 +- * The Hangup Cause family of functions and dialplan applications allow for
 +-   inspection of the hangup cause codes for each channel involved in a call.
 +-   This allows a dialplan writer to determine, for each channel, who hung up and
 +-   for what reason(s).
 +-
 +- * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
 +-   lets you set some of the configuration options from the general section
 +-   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
 +-   the key sequence used to activate built-in features, such as blindxfer,
 +-   and automon.
 +-
 +- * Support for DTLS-SRTP in chan_sip.
 +-
 +- * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
 +-   and callgroups to be defined for several channel drivers.
 +-
 +- * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
 +-
 +- More information about the new features can be found on the Asterisk wiki:
 +-
 +- https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
 +-
 +- A full list of all new features can also be found in the CHANGES file.
 +-
 +- http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
 +-
 +- For a full list of changes in the current release, please see the ChangeLog.
 +-
 +- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2
 +
+ * Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 11.0.0-0.3
+ - fix build on s390
+ 
  * Tue Aug 18 2012 Jeffrey Ollie <jeff at ocjtech.us> - 11.0.0-0.2
  - The Asterisk Development Team is pleased to announce the first beta release of
  - Asterisk 11.0.0.  This release is available for immediate download at
