From 28f8e4aabcd577e5cfb478c5a53170b4af4a2222 Mon Sep 17 00:00:00 2001 From: Jeffrey C. Ollie Date: Oct 30 2012 18:30:41 +0000 Subject: Merge branch 'master' into f18 --- diff --git a/.gitignore b/.gitignore index be43d53..888999b 100644 --- a/.gitignore +++ b/.gitignore @@ -90,3 +90,5 @@ 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 9991396..30b1bee 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 - 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 - 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