- Break up the Misc. section a bit with a new section for Misc. New Modules

- Change spacing a bit in some places for consistent indentation


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -150,11 +150,6 @@ Console Channel Driver changes
   * Added experimental support for video send & receive to chan_oss.
     This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
     a video source.
-  * Added a new channel driver, chan_console, which uses portaudio as a cross
-     platform audio interface.  It was written as a channel driver that would
-     work with Mac CoreAudio, but portaudio supports a number of other audio
-     interfaces, as well. Note that this channel driver requires v19 or higher
-     of portaudio; older versions have a different API.
 
 Phone channel changes (chan_phone)
 ----------------------------------
@@ -202,11 +197,16 @@ Zaptel channel driver (chan_zap) Changes
      state for a channel; also ensured that the DNDState Manager event is
      emitted no matter how the DND state is set or cleared.
 
-A new channel driver: Unistim
------------------------------
+New Channel Drivers
+-------------------
   * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
      configs/unistim.conf.sample for details.  This new channel driver allows
      you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
+  * Added a new channel driver, chan_console, which uses portaudio as a cross
+     platform audio interface.  It was written as a channel driver that would
+     work with Mac CoreAudio, but portaudio supports a number of other audio
+     interfaces, as well. Note that this channel driver requires v19 or higher
+     of portaudio; older versions have a different API.
  
 DUNDi changes
 -------------
@@ -417,6 +417,34 @@ Logger changes
      and to ensure that the oldest log file gets deleted.
   * Added realtime support for the queue log
 
+Miscellaneous New Modules
+-------------------------
+  * Added a new CDR module, cdr_sqlite3_custom.
+  * Added a new realtime configuration module, res_config_sqlite
+  * Added a new codec translation module, codec_resample, which re-samples
+     signed linear audio between 8 kHz and 16 kHz to help support wideband
+     codecs.
+  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
+     based on configuration templates that use Asterisk dialplan function and
+     variable substitution.  It should be possible to create phone profiles and
+     templates that work for the majority of phones provisioned over http. It
+     is currently only intended to provision a single user account per phone.
+     An example profile and set of templates for Polycom phones is provided.
+     NOTE: Polycom firmware is not included, but should be placed in
+     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
+  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
+     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
+     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
+     interfaces create an input and output JACK port.  The application makes
+     these ports the endpoint of the call.  The audio coming from the channel
+     goes out the output port and whatever comes back in on the input port is
+     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
+     audiohook on the channel.  This lets you run the audio coming from a
+     channel through JACK, and whatever comes back in is what gets forwarded
+     on as the channel's audio.  This is very useful for building custom
+     vocoders or doing recording or analysis of the channel's audio in another
+     application.
+
 Miscellaneous 
 -------------
   * Ability to use libcap to set high ToS bits when non-root
@@ -425,10 +453,8 @@ Miscellaneous
   * Added maxfiles option to options section of asterisk.conf which allows you to specify
      what Asterisk should set as the maximum number of open files when it loads.
   * Added the jittertargetextra configuration option.
-  * Added a new CDR module, cdr_sqlite3_custom.
   * The cdr_manager module has a [mappings] feature, like cdr_custom,
     to add fields to the manager event from the CDR variables.
-  * Added a new realtime configuration module, res_config_sqlite
   * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
      configuration files for the IP channel drivers.  The new option is "cos".
      This information is also documented in doc/qos.tex, or the IP Quality of Service
@@ -450,26 +476,4 @@ Miscellaneous
   * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
      specifying which socket to use to connect to the running Asterisk daemon
      (-s)
-  * Added a new codec translation module, codec_resample, which re-samples
-     signed linear audio between 8 kHz and 16 kHz to help support wideband
-     codecs.
-  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
-     based on configuration templates that use Asterisk dialplan function and
-     variable substitution.  It should be possible to create phone profiles and
-     templates that work for the majority of phones provisioned over http. It
-     is currently only intended to provision a single user account per phone.
-     An example profile and set of templates for Polycom phones is provided.
-     NOTE: Polycom firmware is not included, but should be placed in
-     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
-  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
-     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
-     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
-     interfaces create an input and output JACK port.  The application makes
-     these ports the endpoint of the call.  The audio coming from the channel
-     goes out the output port and whatever comes back in on the input port is
-     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
-     audiohook on the channel.  This lets you run the audio coming from a
-     channel through JACK, and whatever comes back in is what gets forwarded
-     on as the channel's audio.  This is very useful for building custom
-     vocoders or doing recording or analysis of the channel's audio in another
-     application.
+