Changes since Asterisk 1.4-beta was branched:
  * Added the bindaddr option to gtalk.conf.
  * Added the ability to specify arguments to the Dial application when using
    the DUNDi switch in the dialplan.
  * 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
  * enable https support for builtin web server.
     See configs/http.conf.sample for details.
  * Argument support for Gosub application
  * MailboxExists converted to dialplan function
  * 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.
  * The output of CallerID in Manager events is now more consistent.
     CallerIDNum is used for number and CallerIDName for name.
  * 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.
  * 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.
  * 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 URI redirect option for the built-in HTTP server
  * 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.

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.