[asterisk] 11.0.0

Tue Oct 30 18:28:31 UTC 2012

commit e2010047ee53422e85ad190f7289b1c1d2765141
Author: Jeffrey C. Ollie <jeff at ocjtech.us>
Date:   Tue Oct 30 13:28:23 2012 -0500

    11.0.0

 .gitignore    |    2 +
 asterisk.spec |   87 +++++++++++++++++++++++++++++++++++++++++++++++++++++++-
 sources       |    4 +-
 3 files changed, 89 insertions(+), 4 deletions(-)
---

diff --git a/.gitignore b/.gitignore
index 9394a48..82fa531 100644
--- a/.gitignore
+++ b/.gitignore
@@ -86,3 +86,5 @@ asterisk-1.8.0-beta3.tar.gz.asc
 /asterisk-11.0.0-rc1.tar.gz.asc
 /asterisk-11.0.0-rc2.tar.gz
 /asterisk-11.0.0-rc2.tar.gz.asc
+/asterisk-11.0.0.tar.gz
+/asterisk-11.0.0.tar.gz.asc
diff --git a/asterisk.spec b/asterisk.spec
index 5022ae4..97a52fc 100644
--- a/asterisk.spec
+++ b/asterisk.spec
@@ -1,4 +1,4 @@
-%global _rc 2
+#global _rc 2
 #global _beta 2
 
 %global _smp_mflags -j1
@@ -31,7 +31,7 @@
 Summary: The Open Source PBX
 Name: asterisk
 Version: 11.0.0
-Release: 0.7%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
+Release: 1%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
 License: GPLv2
 Group: Applications/Internet
 URL: http://www.asterisk.org/
@@ -1384,6 +1384,89 @@ fi
 %{_libdir}/asterisk/modules/app_voicemail_plain.so
 
 %changelog
+* Tue Oct 30 2012 Jeffrey Ollie <jeff at ocjtech.us> - 11.0.0-1:
+- The Asterisk Development Team is pleased to announce the release of
+- Asterisk 11.0.0.  This release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases
+-
+- Asterisk 11 is the next major release series of Asterisk.  It is 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
+
 * Wed Oct 17 2012 Jeffrey Ollie <jeff at ocjtech.us> - 11.0.0-0.7.rc2:
 - The Asterisk Development Team has announced the second release candidate of
 - Asterisk 11.0.0. This release candidate is available for immediate
diff --git a/sources b/sources
index c6604e1..f95e57d 100644
--- a/sources
+++ b/sources
@@ -1,2 +1,2 @@
-04f5fd9b92f9cf79d935cfe5c6962bae  asterisk-11.0.0-rc2.tar.gz
-a0e239ae0131826d948e2c1b7d5c4243  asterisk-11.0.0-rc2.tar.gz.asc
+e23c8535a425253764bdddeee49d1778  asterisk-11.0.0.tar.gz
+16ec07d5f9003044d50175529bd9ca8c  asterisk-11.0.0.tar.gz.asc
