-------------------------------------------------------------------------------
--- Functionality changes since Asterisk 1.4-beta was branched ----------------
-------------------------------------------------------------------------------

Miscellaneous 
-------------

  * Added the bindaddr option to gtalk.conf.
  * Argument support for Gosub application
  * Ability to set process limits without restarting Asterisk
  * SS7 support in chan_zap (via libss7 library)
  * Proper codec support in chan_skinny.
  * AEL upgraded to use the Gosub with Arguments instead
     of Macro application, to hopefully reduce the problems
     seen with the artificially low stack ceiling that 
     Macro bumps into. Macros can only call other Macros
     to a depth of 7. Tests run using gosub, show depths
     limited only by virtual memory. A small test demonstrated
     recursive call depths of 100,000 without problems.
  * Ability to use libcap to set high ToS bits when non-root
     on Linux. If configure is unable to find libcap then you
     can use --with-cap to specify the path.
  * H323 remote hold notification support added (by NOTIFY message
     and/or H.450 supplementary service)
  * Added keepstats option to queues.conf which will keep queue
     statistics during a reload.
  * Added rotatetimestamp option to logger.conf which will use
     the time to name the logger files instead of sequence number.
  * setinterfacevar option in queues.conf also now sets a variable
     called MEMBERNAME which contains the member's name.
  * Added Masquerade manager event for when a masquerade happens between
     two channels.
  * Added 'Strategy' field to manager event QueueParams which represents
     the queue strategy in use. 
  * From the to-do lists: straighten out the app timeout args:
     Wait() app now really does 0.3 seconds- was truncating arg to an int.
     WaitExten() same as Wait().
     Congestion() - Now takes floating pt. argument.
     Busy() - now takes floating pt. argument.
     Read() - timeout now can be floating pt.
     WaitForRing() now takes floating pt timeout arg.
     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
     when kicked out.
  * Added option to run macro when a queue member is connected to a caller, 
     see queues.conf.sample for details.
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
    setqueueentryvar options for each queue, see queues.conf.sample for details.
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added via patch from cfassoni.
  * CID matching information is now shown when doing 'dialplan show'.
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
     does not count paused queue members as unavailable.
  * 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 the trunkmaxsize configuration option to chan_iax2.
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
  * Added the parkedcalltransfers option to features.conf
  * Added 's' option to Page application.
  * Added the srvlookup option to iax.conf
  * Added 'E' and 'V' commands to ExternalIVR.
  * Added 'DBDel' and 'DBDelTree' manager commands.
  * Added 'o' and 'X' options to Chanspy.
  * Added the parkedcallreparking option to features.conf
  * SMDI is now enabled in voicemail using the smdienable option.
  * Added zap show version CLI command to chan_zap.
  * Added a new CDR module, cdr_sqlite3_custom.
  * Added a new realtime configuration module, res_config_sqlite

AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * Added the URI redirect option for the built-in HTTP server
  * The output of CallerID in Manager events is now more consistent.
     CallerIDNum is used for number and CallerIDName for name.
  * enable https support for builtin web server.
     See configs/http.conf.sample for details.
  * Added a new action, GetConfigJSON, which can return the contents of an
    Asterisk configuration file in JSON format.  This is intended to help
    improve the performance of AJAX applications using the manager interface
    over HTTP.

Dialplan functions
------------------
  * Added the DEVSTATE() dialplan function which allows retrieving any device
    state in the dialplan, as well as creating custom device states that are
    controllable from the dialplan.
  * Extend CALLERID() function with "pres" and "ton" parameters to
     fetch string representation of calling number presentation indicator
     and numeric representation of type of calling number value.
  * MailboxExists converted to dialplan function

CLI Changes
-----------
  * New CLI command "core show settings"
  * Added 'core show channels count' CLI command.

SIP changes
-----------
  * The default SIP useragent= identifier now includes the Asterisk version
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
     If set, and the incoming request carries authentication info,
     the username to match in the users list is taken from the Digest header
     rather than from the From: field. This feature is considered experimental.
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
  * The "localmask" setting was removed in version 1.2 and the reminder about it
     being removed is now also removed.
  * A new option "busy-level" for setting a level of calls where asterisk reports
     a device as busy, to separate it from call-limit
  * A new realtime family called "sipregs" is now supported to store SIP registration
     data. If this family is defined, "sippeers" will be used for configuration and
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
     registration data, as before.
  * The SIPPEER function have new options for port address, call and pickup groups
  * Added support for T.140 realtime text in SIP/RTP

DUNDi changes
-------------
  * Added the ability to specify arguments to the Dial application when using
     the DUNDi switch in the dialplan.
  * Added the ability to set weights for responses dynamically.  This can be
     done using a global variable or a dialplan function.  Using the SHELL()
     function would allow you to have an external script set the weight for
     each response.

Voicemail Changes
-----------------
  * Added the ability to customize which sound files are used for some of the
     prompts within the Voicemail application by changing them in voicemail.conf
  * Added the ability for the "voicemail show users" CLI command to show users
    configured by the dynamic realtime configuration method.
