From 4dddb6fa47cbd718b19e50b88a39db0010917ec6 Mon Sep 17 00:00:00 2001 From: Jeffrey C. Ollie Date: Sep 26 2012 16:32:19 +0000 Subject: 11.0.0-beta2 --- diff --git a/.gitignore b/.gitignore index 4c12294..7fac667 100644 --- a/.gitignore +++ b/.gitignore @@ -80,3 +80,5 @@ asterisk-1.8.0-beta3.tar.gz.asc /asterisk-10.5.2.tar.gz.asc /asterisk-11.0.0-beta1.tar.gz /asterisk-11.0.0-beta1.tar.gz.asc +/asterisk-11.0.0-beta2.tar.gz +/asterisk-11.0.0-beta2.tar.gz.asc diff --git a/asterisk.spec b/asterisk.spec index f3d3516..ddd2b40 100644 --- a/asterisk.spec +++ b/asterisk.spec @@ -1,5 +1,5 @@ #global _rc 3 -%global _beta 1 +%global _beta 2 %global _smp_mflags -j1 @@ -31,7 +31,7 @@ Summary: The Open Source PBX Name: asterisk Version: 11.0.0 -Release: 0.2%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist} +Release: 0.3%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist} License: GPLv2 Group: Applications/Internet URL: http://www.asterisk.org/ @@ -84,6 +84,7 @@ BuildRequires: latex2html # for building res_calendar_caldav BuildRequires: neon-devel%{?_isa} BuildRequires: libical-devel%{?_isa} +BuildRequires: libxml2-devel%{?_isa} # for codec_speex BuildRequires: speex-devel%{?_isa} >= 1.2 @@ -102,6 +103,9 @@ BuildRequires: SDL_image-devel%{?_isa} # cli BuildRequires: libedit-devel%{?_isa} +# codec_ilbc +BuildRequires: ilbc-devel%{?_isa} + Requires(pre): %{_sbindir}/useradd Requires(pre): %{_sbindir}/groupadd %if %{systemd} @@ -576,9 +580,9 @@ pushd menuselect popd %if 0%{?fedora} > 0 -%configure --host=%{_target_platform} --with-imap=system --with-gsm=/usr --with-libedit=yes --with-srtp LDFLAGS="%{ldflags}" +%configure --host=%{_target_platform} --with-imap=system --with-gsm=/usr --with-ilbc=/usr --with-libedit=yes --with-srtp LDFLAGS="%{ldflags}" %else -%configure --host=%{_target_platform} --with-gsm=/usr --with-libedit=yes --with-gmime=no --with-srtp LDFLAGS="%{ldflags}" +%configure --host=%{_target_platform} --with-gsm=/usr --with-ilbc=/usr --with-libedit=yes --with-gmime=no --with-srtp LDFLAGS="%{ldflags}" %endif ASTCFLAGS="%{optflags}" LDFLAGS="%{ldflags}" make %{?_smp_mflags} menuselect-tree @@ -888,6 +892,7 @@ fi %{_libdir}/asterisk/modules/codec_g722.so %{_libdir}/asterisk/modules/codec_g726.so %{_libdir}/asterisk/modules/codec_gsm.so +%{_libdir}/asterisk/modules/codec_ilbc.so %{_libdir}/asterisk/modules/codec_lpc10.so %{_libdir}/asterisk/modules/codec_resample.so %{_libdir}/asterisk/modules/codec_speex.so @@ -1374,6 +1379,96 @@ fi %{_libdir}/asterisk/modules/app_voicemail_plain.so %changelog +* Wed Sep 26 2012 Jeffrey Ollie - 11.0.0-0.3 +- 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 Aug 18 2012 Jeffrey Ollie - 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 diff --git a/menuselect.makeopts b/menuselect.makeopts index 376c57f..1eb4634 100644 --- a/menuselect.makeopts +++ b/menuselect.makeopts @@ -4,7 +4,7 @@ MENUSELECT_BRIDGES= MENUSELECT_CDR=cdr_sqlite MENUSELECT_CFLAGS=LOADABLE_MODULES MENUSELECT_CHANNELS=chan_h323 chan_nbs chan_vpb -MENUSELECT_CODECS=codec_ilbc +MENUSELECT_CODECS= MENUSELECT_CORE_SOUNDS= MENUSELECT_EMBED= MENUSELECT_EXTRA_SOUNDS= diff --git a/sources b/sources index 4eb3f3c..ec96feb 100644 --- a/sources +++ b/sources @@ -1,2 +1,2 @@ -a99bdeae82f80b25b093eba957581dd9 asterisk-11.0.0-beta1.tar.gz -f9a6f1e27d61287f9b32464911d9c98d asterisk-11.0.0-beta1.tar.gz.asc +b04db6d6828748c73bc5ae7f5c1dc3e6 asterisk-11.0.0-beta2.tar.gz +53836a4190292c4fcf6ccbd88ef2e556 asterisk-11.0.0-beta2.tar.gz.asc