UPGRADE.txt

  Previous to 1.6, macros were implemented with the Macro() app, which
  provided a nice feature of auto-returning. The compiler will do its
  best to insert a Return() app call at the end of your macro if you did
  not include it, but really, you should make sure that all execution
  paths within your macros end in "return;".

* The conf2ael program is 'introduced' in this release; it is in a rather
  crude state, but deemed useful for making a first pass at converting
  extensions.conf code into AEL. More intelligence will come with time.

Core:

* The 'languageprefix' option in asterisk.conf is now deprecated, and
  the default sound file layout for non-English sounds is the 'new
  style' layout introduced in Asterisk 1.4 (and used by the automatic
  sound file installer in the Makefile).

* The ast_expr2 stuff has been modified to handle floating-point numbers.
  Numbers of the format D.D are now acceptable input for the expr parser,
  Where D is a string of base-10 digits. All math is now done in "long double",
  if it is available on your compiler/architecture. This was half-way between
  a bug-fix (because the MATH func returns fp by default), and an enhancement.
  Also, for those counting on, or needing, integer operations, a series of
  'functions' were also added to the expr language, to allow several styles
  of rounding/truncation, along with a set of common floating point operations,
  like sin, cos, tan, log, pow, etc. The ability to call external functions
  like CDR(), etc. was also added, without having to use the ${...} notation.

* The delimiter passed to applications has been changed to the comma (','), as
  that is what people are used to using within extensions.conf.  If you are
  using realtime extensions, you will need to translate your existing dialplan
  to use this separator.  To use a literal comma, you need merely to escape it
  with a backslash ('\').  Another possible side effect is that you may need to
  remove the obscene level of backslashing that was necessary for the dialplan
  to work correctly in 1.4 and previous versions.  This should make writing
  dialplans less painful in the future, albeit with the pain of a one-time
  conversion.  If you would like to avoid this conversion immediately, set
  pbx_realtime=1.4 in the [compat] section of asterisk.conf.  After
  transitioning, set pbx_realtime=1.6 in the same section.

* For the same purpose as above, you may set res_agi=1.4 in the [compat]
  section of asterisk.conf to continue to use the '|' delimiter in the EXEC
  arguments of AGI applications.  After converting to use the ',' delimiter,
  change this option to res_agi=1.6.

* As a side effect of the application delimiter change, many places that used
  to need quotes in order to get the proper meaning are no longer required.
  You now only need to quote strings in configuration files if you literally
  want quotation marks within a string.

* Any applications run that contain the pipe symbol but not a comma symbol will
  get a warning printed to the effect that the application delimiter has changed.
  However, there are legitimate reasons why this might be useful in certain
  situations, so this warning can be turned off with the dontwarn option in
  asterisk.conf.

* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
  'rotatestrategy'.  This new option supports a 'rotate' strategy that more
  closely mimics the system logger in terms of file rotation.

* The concise versions of various CLI commands are now deprecated. We recommend
  using the manager interface (AMI) for application integration with Asterisk.

Voicemail:

* The voicemail configuration values 'maxmessage' and 'minmessage' have
  been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
  to make them more distinguishable from 'maxmsgs', which sets folder
  size.  The old variables will continue to work in this version, albeit
  with a deprecation warning.

* If you use any interface for modifying voicemail aside from the built in
  dialplan applications, then the option "pollmailboxes" *must* be set in
  voicemail.conf for message waiting indication (MWI) to work properly.  This
  is because Voicemail notification is now event based instead of polling
  based.  The channel drivers are no longer responsible for constantly manually
  checking mailboxes for changes so that they can send MWI information to users.
  Examples of situations that would require this option are web interfaces to
  voicemail or an email client in the case of using IMAP storage.

Applications:


* ChanIsAvail() now has a 't' option, which allows the specified device
  to be queried for state without consulting the channel drivers. This
  performs mostly a 'ChanExists' sort of function.

* ChannelRedirect() will not terminate the channel that fails to do a
  channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
  will reflect if the attempt was successful of not.

* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
  and is now deprecated.

* DISA()'s fifth argument is now an options argument.  If you have previously
  used 'NOANSWER' in this argument, you'll need to convert that to the new
  option 'n'.

* Macro() is now deprecated.  If you need subroutines, you should use the
  Gosub()/Return() applications.  To replace MacroExclusive(), we have
  introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK().  You may use
  these functions in any location where you desire to ensure that only one
  channel is executing that path at any one time.  The Macro() applications
  are deprecated for performance reasons.  However, since Macro() has been
  around for a long time and so many dialplans depend heavily on it, for the
  sake of backwards compatibility it will not be removed .  It is also worth
  noting that using both Macro() and GoSub() at the same time is _heavily_
  discouraged.

* Read() now sets a READSTATUS variable on exit.  It does NOT automatically
  return -1 (and hangup) anymore on error.  If you want to hangup on error,
  you need to do so explicitly in your dialplan.

* Privacy() no longer uses privacy.conf, so any options must be specified
  directly in the application arguments.

* MusicOnHold application now has duration parameter which allows specifying
  timeout in seconds.

* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.

* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
  instead.

* The arguments in ExecIf changed a bit, to be more like other applications.
  The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).

* The behavior of the Set application now depends upon a compatibility option,
  set in asterisk.conf.  To use the old 1.4 behavior, which allowed Set to take
  multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf.  To
  use the new behavior, which permits variables to be set with embedded commas,
  set app_set=1.6 in [compat] in asterisk.conf.  Note that you can have both
  behaviors at the same time, if you switch to using MSet if you want the old
  behavior.

Dialplan Functions:

* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
  more information, issue a "show function QUEUE_MEMBER" from the CLI.

CDR:

* The cdr_sqlite module has been marked as deprecated in favor of
  cdr_sqlite3_custom.  It will potentially be removed from the tree
  after Asterisk 1.6 is released.

* The cdr_odbc module now uses res_odbc to manage its connections.  The
  username and password parameters in cdr_odbc.conf, therefore, are no
  longer used.  The dsn parameter now points to an entry in res_odbc.conf.

* The uniqueid field in the core Asterisk structure has been changed from a
  maximum 31 character field to a 149 character field, to account for all
  possible values the systemname prefix could be.  In the past, if the
  systemname was too long, the uniqueid would have been truncated.

* The cdr_tds module now supports all versions of FreeTDS that contain
  the db-lib frontend.  It will also now log the userfield variable if
  the target database table contains a column for it.

Formats:

* format_wav: The GAIN preprocessor definition and source code that used it
  is removed.  This change was made in response to user complaints of
  choppiness or the clipping of loud signal peaks.  To increase the volume
  of voicemail messages, use the 'volgain' option in voicemail.conf

Channel Drivers:

* SIP: a small upgrade to support the "Record" button on the SNOM360,
  which sends a sip INFO message with a "Record: on" or "Record: off"
  header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
  requests (by default, via '*1'), then the user-configured dialpad sequence
  is generated, and recording can be started and stopped via this button. The
  file names and formats are all controlled via the normal mechanisms. If the
  user has not configured the automon feature, the normal "415 Unsupported media type"
  is returned, and nothing is done.

* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
  Asterisk, but will be removed in the following version. Please use the groupcount functions
  in the dialplan to enforce call limits. The "limitonpeer" configuration option is
  now renamed to "counteronpeer".

* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
  These are used only before registration to call a peer with the uri
	sip:defaultuser@defaultip
  The "username" setting still work, but is deprecated and will not work in
  the next version of Asterisk.

* SIP: The old "insecure" options, deprecated in 1.4, have been removed.
  "insecure=very" should be changed to "insecure=port,invite"
  "insecure=yes" should be changed to "insecure=port"
  Be aware that some telephony providers show the invalid syntax in their
  sample configurations.

* chan_local.c: the comma delimiter inside the channel name has been changed to a
  semicolon, in order to make the Local channel driver compatible with the comma
  delimiter change in applications.

* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
  to be compatible with settings in sip.conf. The "tos" and "cos" configuration
  is deprecated and will stop working in the next release of Asterisk.

* Console: A new console channel driver, chan_console, has been added to Asterisk.
  This new module can not be loaded at the same time as chan_alsa or chan_oss.  The
  default modules.conf only loads one of them (chan_oss by default).  So, unless you
  have modified your modules.conf to not use the autoload option, then you will need
  to modify modules.conf to add another "noload" line to ensure that only one of
  these three modules gets loaded.

* DAHDI: The chan_zap module that supported PSTN interfaces using
  Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
  telephony driver package for PSTN interfaces. See the
  Zaptel-to-DAHDI.txt file for more details on this transition.

* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
  the method of stripping digits in the dialplan using variable substring syntax.

Configuration:

* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
  lowcost and other is not acceptable now. Look into qos.tex for description of
  this parameter.

* queues.conf: the queue-lessthan sound file option is no longer available, and the
  queue-round-seconds option no longer takes '1' as a valid parameter.

Manager:

* Manager has been upgraded to version 1.1 with a lot of changes.
  Please check doc/manager_1_1.txt for information

* The IAXpeers command output has been changed to more closely resemble the
  output of the SIPpeers command.

* cdr_manager now reports at the "cdr" level, not at "call"  You may need to
   change your manager.conf to add the level to existing AMI users, if they
   want to see the CDR events generated.

* The Originate command now requires the Originate write permission.  For
   Originate with the Application parameter, you need the additional System
   privilege if you want to do anything that calls out to a subshell.

iLBC Codec:

* Previously, the Asterisk source code distribution included the iLBC
  encoder/decoder source code, from Global IP Solutions
  (http://www.gipscorp.com). This code is not licensed for
  distribution, and thus has been removed from the Asterisk source
  code distribution. If you wish to use codec_ilbc to support iLBC
  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
  script to download the source and put it in the proper place in
  the Asterisk build tree. Once that is done you can follow your normal
  steps of building Asterisk. You will need to run 'menuselect' and enable
  the iLBC codec in the 'Codec  Translators' category.

From 1.2 to 1.4:

Build Process (configure script):

Asterisk now uses an autoconf-generated configuration script to learn how it
should build itself for your system. As it is a standard script, running:

$ ./configure --help

will show you all the options available. This script can be used to tell the
build process what libraries you have on your system (if it cannot find them
automatically), which libraries you wish to have ignored even though they may
be present, etc.

You must run the configure script before Asterisk will build, although it will
attempt to automatically run it for you with no options specified; for most
users, that will result in a similar build to what they would have had before
the configure script was added to the build process (except for having to run
'make' again after the configure script is run). Note that the configure script
does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
when your system configuration changes or you wish to build Asterisk with
different options.

Build Process (module selection):

The Asterisk source tree now includes a basic module selection and build option
selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
In this tool, you can disable building of modules that you don't care about,
turn on/off global options for the build and see which modules will not
(and cannot) be built because your system does not have the required external
dependencies installed.

The resulting file from menuselect is called 'menuselect.makeopts'. Note that
the resulting menuselect.makeopts file generally contains which modules *not*
to build. The modules listed in this file indicate which modules have unmet
dependencies, a present conflict, or have been disabled by the user in the
menuselect interface. Compiler Flags can also be set in the menuselect
interface.  In this case, the resulting file contains which CFLAGS are in use,
not which ones are not in use.

If you would like to save your choices and have them applied against all
builds, the file can be copied to '~/.asterisk.makeopts' or
'/etc/asterisk.makeopts'.

Build Process (Makefile targets):

The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
in the menuselect tool.

It is now possible to run most make targets against a single subdirectory; from
the top level directory, for example, 'make channels' will run 'make all' in the
'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.

Sound (prompt) and Music On Hold files:

Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
use with Asterisk have been replaced with new versions produced from high quality
master recordings, and are available in three languages (English, French and
Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
In addition, the music on hold files provided by opsound.org Music are now available
in the same five formats, but no longer available in MP3 format.

The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
All of the other variations can be installed by running 'make menuselect' and
selecting the packages you wish to install; when you run 'make install', those
packages will be downloaded and installed along with the standard files included
in the tarball.

If for some reason you expect to not have Internet access at the time you will be
running 'make install', you can make your package selections using menuselect and
then run 'make sounds' to download (only) the sound packages; this will leave the
sound packages in the 'sounds' subdirectory to be used later during installation.

WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
instead of the alternate-language files being stored in subdirectories underneath
the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
language itself, then places all the sound files for that language under that
directory and its subdirectories. This is the layout that will be created if you
select non-English languages to be installed via menuselect, HOWEVER Asterisk does
not default to this layout and will not find the files in the places it expects them
to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
installed.

PBX Core:

* The (very old and undocumented) ability to use BYEXTENSION for dialing
  instead of ${EXTEN} has been removed.

* Builtin (res_features) transfer functionality attempts to use the context
  defined in TRANSFER_CONTEXT variable of the transferer channel first. If
  not set, it uses the transferee variable. If not set in any channel, it will
  attempt to use the last non macro context. If not possible, it will default
  to the current context.

* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
  if your dialplan relies on the ability to 'run off the end' of an extension
  and wait for a new extension without using WaitExten() to accomplish that,
  you will need set autofallthrough to 'no' in your extensions.conf file.

Command Line Interface:

* 'show channels concise', designed to be used by applications that will parse
  its output, previously used ':' characters to separate fields. However, some
  of those fields can easily contain that character, making the output not
  parseable. The delimiter has been changed to '!'.

Applications:

* In previous Asterisk releases, many applications would jump to priority n+101
  to indicate some kind of status or error condition.  This functionality was
  marked deprecated in Asterisk 1.2.  An option to disable it was provided with
  the default value set to 'on'.  The default value for the global priority
  jumping option is now 'off'.

* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
  AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
  and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
  been removed in this version.  You should use the equivalent dialplan
  function in places where you have previously used one of these applications.

* The application SetGlobalVar has been deprecated.  You should replace uses
  of this application with the following combination of Set and GLOBAL():
  Set(GLOBAL(name)=value).  You may also access global variables exclusively by
  using the GLOBAL() dialplan function, instead of relying on variable
  interpolation falling back to globals when no channel variable is set.

* The application SetVar has been renamed to Set.  The syntax SetVar was marked
  deprecated in version 1.2 and is no longer recognized in this version.  The
  use of Set with multiple argument pairs has also been deprecated.  Please
  separate each name/value pair into its own dialplan line.

* app_read has been updated to use the newer options codes, using "skip" or
  "noanswer" will not work.  Use s or n.  Also there is a new feature i, for
  using indication tones, so typing in skip would give you unexpected results.

* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.

* The CONNECT event in the queue_log from app_queue now has a second field
  in addition to the holdtime field. It contains the unique ID of the
  queue member channel that is taking the call. This is useful when trying
  to link recording filenames back to a particular call from the queue.

* The old/current behavior of app_queue has a serial type behavior
  in that the queue will make all waiting callers wait in the queue
  even if there is more than one available member ready to take
  calls until the head caller is connected with the member they
  were trying to get to. The next waiting caller in line then
  becomes the head caller, and they are then connected with the
  next available member and all available members and waiting callers
  waits while this happens. This cycle continues until there are
  no more available members or waiting callers, whichever comes first.
  The new behavior, enabled by setting autofill=yes in queues.conf
  either at the [general] level to default for all queues or
  to set on a per-queue level, makes sure that when the waiting
  callers are connecting with available members in a parallel fashion
  until there are no more available members or no more waiting callers,
  whichever comes first. This is probably more along the lines of how
  one would expect a queue should work and in most cases, you will want
  to enable this new behavior. If you do not specify or comment out this
  option, it will default to "no" to keep backward compatability with the old
  behavior.

* Queues depend on the channel driver reporting the proper state
  for each member of the queue. To get proper signalling on
  queue members that use the SIP channel driver, you need to
  enable a call limit (could be set to a high value so it
  is not put into action) and also make sure that both inbound
  and outbound calls are accounted for.

  Example:

       [general]
       limitonpeer = yes

       [peername]
       type=friend
       call-limit=10


* The app_queue application now has the ability to use MixMonitor to
  record conversations queue members are having with queue callers. Please
  see configs/queues.conf.sample for more information on this option.

* The app_queue application strategy called 'roundrobin' has been deprecated
  for this release. Users are encouraged to use 'rrmemory' instead, since it
  provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
  'rrmemory' will be renamed 'roundrobin'.

* The app_queue application option called 'monitor-join' has been deprecated
  for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
  since it provides the same functionality but is not dependent on soxmix or some
  other external program in order to mix the audio.

* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
  the 'm' option now provides the functionality of "initially muted".
  In practice, most existing dialplans using the 'm' flag should not notice
  any difference, unless the keypad menu is enabled, allowing the user
  to unmute themselves.

* ast_play_and_record would attempt to cancel the recording if a DTMF
  '0' was received.  This behavior was not documented in most of the
  applications that used ast_play_and_record and the return codes from
  ast_play_and_record weren't checked for properly.
  ast_play_and_record has been changed so that '0' no longer cancels a
  recording.  If you want to allow DTMF digits to cancel an
  in-progress recording use ast_play_and_record_full which allows you
  to specify which DTMF digits can be used to accept a recording and
  which digits can be used to cancel a recording.

* ast_app_messagecount has been renamed to ast_app_inboxcount.  There is now a
  new ast_app_messagecount function which takes a single context/mailbox/folder
  mailbox specification and returns the message count for that folder only.
  This addresses the deficiency of not being able to count the number of
  messages in folders other than INBOX and Old.

* The exit behavior of the AGI applications has changed. Previously, when
  a connection to an AGI server failed, the application would cause the channel
  to immediately stop dialplan execution and hangup. Now, the only time that
  the AGI applications will cause the channel to stop dialplan execution is
  when the channel itself requests hangup. The AGI applications now set an
  AGISTATUS variable which will allow you to find out whether running the AGI
  was successful or not.

  Previously, there was no way to handle the case where Asterisk was unable to
  locally execute an AGI script for some reason. In this case, dialplan
  execution will continue as it did before, but the AGISTATUS variable will be
  set to "FAILURE".

  A locally executed AGI script can now exit with a non-zero exit code and this
  failure will be detected by Asterisk. If an AGI script exits with a non-zero
  exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
  "SUCCESS".

* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
  previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
  your table format using the schema provided in doc/odbcstorage.txt

* app_waitforsilence: Fixes have been made to this application which changes the
  default behavior with how quickly it returns. You can maintain "old-style" behavior
  with the addition/use of a third "timeout" parameter.
  Please consult the application documentation and make changes to your dialplan
  if appropriate.

Manager:

* After executing the 'status' manager action, the "Status" manager events
  included the header "CallerID:" which was actually only the CallerID number,
  and not the full CallerID string.  This header has been renamed to
  "CallerIDNum".  For compatibility purposes, the CallerID parameter will remain
  until after the release of 1.4, when it will be removed.  Please use the time
  during the 1.4 release to make this transition.

* The AgentConnect event now has an additional field called "BridgedChannel"
  which contains the unique ID of the queue member channel that is taking the
  call. This is useful when trying to link recording filenames back to
  a particular call from the queue.

* app_userevent has been modified to always send Event: UserEvent with the
  additional header UserEvent: <userspec>.  Also, the Channel and UniqueID
  headers are not automatically sent, unless you specify them as separate
  arguments.  Please see the application help for the new syntax.

* app_meetme: Mute and Unmute events are now reported via the Manager API.
  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
  are easier to use than "Action Command:". The MeetMeStopTalking event has
  also been deprecated in favor of the already existing MeetmeTalking event
  with a "Status" of "on" or "off" added.

* OriginateFailure and OriginateSuccess events were replaced by event
  OriginateResponse with a header named "Response" to indicate success or
  failure

Variables:

* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
  ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
  and ${LANGUAGE} have all been deprecated in favor of their related dialplan
  functions.  You are encouraged to move towards the associated dialplan
  function, as these variables will be removed in a future release.

* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
  adjustable from cdr.conf, instead of recompiling.

* OSP applications exports several new variables, ${OSPINHANDLE},
  ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
  ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}

* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
  created channel. This variables holds the channel name of the transferer.

* The dial plan variable PRI_CAUSE will be removed from future versions
  of Asterisk.
  It is replaced by adding a cause value to the hangup() application.

Functions:

* The function ${CHECK_MD5()} has been deprecated in favor of using an
  expression: $[${MD5(<string>)} = ${saved_md5}].

* The 'builtin' functions that used to be combined in pbx_functions.so are
  now built as separate modules. If you are not using 'autoload=yes' in your
  modules.conf file then you will need to explicitly load the modules that
  contain the functions you want to use.

* The ENUMLOOKUP() function with the 'c' option (for counting the number of
  records), but the lookup fails to match any records, the returned value will
  now be "0" instead of blank.

* The REALTIME() function is now available in version 1.4 and app_realtime has
  been deprecated in favor of the new function. app_realtime will be removed
  completely with the version 1.6 release so please take the time between
  releases to make any necessary changes

* The QUEUEAGENTCOUNT() function has been deprecated in favor of
  QUEUE_MEMBER_COUNT().

The IAX2 channel:

* It is possible that previous configurations depended on the order in which
  peers and users were specified in iax.conf for forcing the order in which
  chan_iax2 matched against them.  This behavior is going away and is considered
  deprecated in this version.  Avoid having ambiguous peer and user entries and
  to make things easy on yourself, always set the "username" option for users
  so that the remote end can match on that exactly instead of trying to infer
  which user you want based on host.

  If you would like to go ahead and use the new behavior which doesn't use the
  order in the config file to influence matching order, then change the
  MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one.  An
  example is provided there.  By changing this, you will get *much* better
  performance on systems that do a lot of peer and user lookups as they will be
  stored in memory in a much more efficient manner.

* The "mailboxdetail" option has been deprecated.  Previously, if this option
  was not enabled, the 2 byte MSGCOUNT information element would be set to all
  1's to indicate there there is some number of messages waiting.  With this
  option enabled, the number of new messages were placed in one byte and the
  number of old messages are placed in the other.  This is now the default
  (and the only) behavior.

The SIP channel:

* The "incominglimit" setting is replaced by the "call-limit" setting in
  sip.conf.

* OSP support code is removed from SIP channel to OSP applications. ospauth
  option in sip.conf is removed to osp.conf as authpolicy. allowguest option
  in sip.conf cannot be set as osp anymore.

* The Asterisk RTP stack has been changed in regards to RFC2833 reception
  and transmission. Packets will now be sent with proper duration instead of all
  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
  will want to turn on the rfc2833compensate option. Without this option your
  DTMF reception may act poorly.

* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
  in coming versions of Asterisk. Please use the dialplan function
  SIPCHANINFO(useragent) instead.

* The ALERT_INFO dialplan variable is deprecated and will be removed
  in coming versions of Asterisk. Please use the dialplan application
  sipaddheader() to add the "Alert-Info" header to the outbound invite.

* The "canreinvite" option has changed. canreinvite=yes used to disable
  re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
  to disable re-invites when NAT=yes. This is propably what you want.
  The settings are now: "yes", "no", "nonat", "update". Please consult
  sip.conf.sample for detailed information.

The Zap channel:

* Support for MFC/R2 has been removed, as it has not been functional for some
  time and it has no maintainer.

The Agent channel:

* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
  it provided can be done using dialplan logic, without requiring additional
  channel and module locks (which frequently caused deadlocks). An example of
  how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.

The G726-32 codec:

* It has been determined that previous versions of Asterisk used the wrong codeword
  packing order for G726-32 data. This version supports both available packing orders,
  and can transcode between them. It also now selects the proper order when
  negotiating with a SIP peer based on the codec name supplied in the SDP. However,
  there are existing devices that improperly request one order and then use another;
  Sipura and Grandstream ATAs are known to do this, and there may be others. To
  be able to continue to use these devices with this version of Asterisk and the
  G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
  to sip.conf, so that Asterisk can use the packing order expected by the device (even
  though it requested a different order). In addition, the internal format number for
  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
  result of this is that this version of Asterisk will be able to interoperate over
  IAX2 with older versions of Asterisk, as long as this version is told to allow
  'g726aal2' instead of 'g726' as the codec for the call.

Installation:

* On BSD systems, the installation directories have changed to more "FreeBSDish"
  directories. On startup, Asterisk will look for the main configuration in
  /usr/local/etc/asterisk/asterisk.conf
  If you have an old installation, you might want to remove the binaries and
  move the configuration files to the new locations. The following directories
  are now default:
	ASTLIBDIR	/usr/local/lib/asterisk
	ASTVARLIBDIR	/usr/local/share/asterisk
	ASTETCDIR	/usr/local/etc/asterisk
	ASTBINDIR	/usr/local/bin/asterisk
	ASTSBINDIR	/usr/local/sbin/asterisk

Music on Hold:

* The music on hold handling has been changed in some significant ways in hopes
  to make it work in a way that is much less confusing to users. Behavior will
  not change if the same configuration is used from older versions of Asterisk.
  However, there are some new configuration options that will make things work
  in a way that makes more sense.

  Previously, many of the channel drivers had an option called "musicclass" or
  something similar. This option set what music on hold class this channel
  would *hear* when put on hold. Some people expected (with good reason) that
  this option was to configure what music on hold class to play when putting
  the bridged channel on hold. This option has now been deprecated.

  Two new music on hold related configuration options for channel drivers have
  been introduced. Some channel drivers support both options, some just one,
  and some support neither of them. Check the sample configuration files to see
  which options apply to which channel driver.

  The "mohsuggest" option specifies which music on hold class to suggest to the
  bridged channel when putting them on hold. The only way that this class can
  be overridden is if the bridged channel has a specific music class set that
  was done in the dialplan using Set(CHANNEL(musicclass)=something).

  The "mohinterpret" option is similar to the old "musicclass" option. It
  specifies which music on hold class this channel would like to listen to when
  put on hold. This music class is only effective if this channel has no music
  class set on it from the dialplan and the bridged channel putting this one on
  hold had no "mohsuggest" setting.

  The IAX2 and Zap channel drivers have an additional feature for the
  "mohinterpret" option. If this option is set to "passthrough", then these
  channel drivers will pass through the HOLD message in signalling instead of
  starting music on hold on the channel. An example for how this would be
  useful is in an enterprise network of Asterisk servers. When one phone on one
  server puts a phone on a different server on hold, the remote server will be
  responsible for playing the hold music to its local phone that was put on
  hold instead of the far end server across the network playing the music.

CDR Records:

* The behavior of the "clid" field of the CDR has always been that it will
  contain the callerid ANI if it is set, or the callerid number if ANI was not
  set.  When using the "callerid" option for various channel drivers, some
  would set ANI and some would not.  This has been cleared up so that all
  channel drivers set ANI.  If you would like to change the callerid number
  on the channel from the dialplan and have that change also show up in the
  CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).

API:

* There are some API functions that were not previously prefixed with the 'ast_'
  prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
  have a module that uses the services provided by res_adsi, res_odbc, or
  res_agi, you will need to add ast_ prefixes to the functions that you call
  from those modules.

Formats:

* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
  in Asterisk 1.4.  This change was made in response to user complaints of
  choppiness or the clipping of loud signal peaks.  The GAIN preprocessor
  definition will be retained in Asterisk 1.4, but will be removed in a
  future release.  The use of GAIN for the increasing of voicemail message
  volume should use the 'volgain' option in voicemail.conf

From 1.0 to 1.2:

Compiling:

* The Asterisk 1.2 source code now uses C language features
  supported only by 'modern' C compilers.  Generally, this means GCC
  version 3.0 or higher, although some GCC 2.96 releases will also
  work.  Some non-GCC compilers that support C99 and the common GCC
  extensions (including anonymous structures and unions) will also
  work.  All releases of GCC 2.95 do _not_ have the requisite feature
  support; systems using that compiler will need to be upgraded to
  a more recent compiler release.

Dialplan Expressions:

* The dialplan expression parser (which handles $[ ... ] constructs)
  has gone through a major upgrade, but has one incompatible change:
  spaces are no longer required around expression operators, including
  string comparisons. However, you can now use quoting to keep strings
  together for comparison. For more details, please read the
  doc/README.variables file, and check over your dialplan for possible
  problems.

Agents:

* The default for ackcall has been changed to "no" instead of "yes"
  because of a bug which caused the "yes" behavior to generally act like
  "no".  You may need to adjust the value if your agents behave
  differently than you expect with respect to acknowledgement.

* The AgentCallBackLogin application now requires a second '|' before
  specifying an extension@context.  This is to distinguish the options
  string from the extension, so that they do not conflict.  See
  'show application AgentCallbackLogin' for more details.

Parking:

* Parking behavior has changed slightly; when a parked call times out,
  Asterisk will attempt to deliver the call back to the extension that
  parked it, rather than the 's' extension. If that extension is busy
  or unavailable, the parked call will be lost.

Dialing:

* The Caller*ID of the outbound leg is now the extension that was
  called, rather than the Caller*ID of the inbound leg of the call.  The
  "o" flag for Dial can be used to restore the original behavior if
  desired.  Note that if you are looking for the originating callerid
  from the manager event, there is a new manager event "Dial" which
  provides the source and destination channels and callerid.

IAX:

* The naming convention for IAX channels has changed in two ways:
   1. The call number follows a "-" rather than a "/" character.
   2. The name of the channel has been simplified to IAX2/peer-callno,
   rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.

SIP:

* The global option "port" in 1.0.X that is used to set which port to
  bind to has been changed to "bindport" to be more consistent with
  the other channel drivers and to avoid confusion with the "port"
  option for users/peers.

* The "Registry" event now uses "Username" rather than "User" for
  consistency with IAX.

Applications:

* With the addition of dialplan functions (which operate similarly
  to variables), the SetVar application has been renamed to Set.

* The CallerPres application has been removed.  Use SetCallerPres
  instead.  It accepts both numeric and symbolic names.

* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
  CheckGroup have been deprecated in favor of functions.  Here is a
  table of their replacements:

  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)

  Note that CheckGroup does not have a direct replacement.  There is
  also a new function called GROUP_LIST() which will return a space
  separated list of all of the groups set on a channel.  The GROUP()
  function can also return the name of the group set on a channel when
  used in a read environment.

* The applications DBGet and DBPut have been deprecated in favor of
  functions.  Here is a table of their replacements:

  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})

* The application SetLanguage has been deprecated in favor of the
  function LANGUAGE().

  SetLanguage(fr)		Set(LANGUAGE()=fr)

  The LANGUAGE function can also return the currently set language:

  Set(MYLANG=${LANGUAGE()})

* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
  have been deprecated in favor of the function TIMEOUT(timeouttype):

  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
  ResponseTimeout(15)		Set(TIMEOUT(response)=15)

  The TIMEOUT() function can also return the currently set timeouts:

  Set(DTIMEOUT=${TIMEOUT(digit)})

* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
  deprecated in favor of the CALLERID(datatype) function:

  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)

* The application Record now uses the period to separate the filename
  from the format, rather than the colon.

* The application VoiceMail now supports a 'temporary' greeting for each
  mailbox. This greeting can be recorded by using option 4 in the
  'mailbox options' menu, and 'change your password' option has been
  moved to option 5.

* The application VoiceMailMain now only matches the 'default' context if
  none is specified in the arguments.  (This was the previously
  documented behavior, however, we didn't follow that behavior.)  The old
  behavior can be restored by setting searchcontexts=yes in voicemail.conf.

Queues:

* A queue is now considered empty not only if there are no members but if
  none of the members are available (e.g. agents not logged on).  To
  restore the original behavior, use "leavewhenempty=strict" or
  "joinwhenempty=strict" instead of "=yes" for those options.

* It is now possible to use multi-digit extensions in the exit context
  for a queue (although you should not have overlapping extensions,
  as there is no digit timeout). This means that the EXITWITHKEY event
  in queue_log can now contain a key field with more than a single
  character in it.

Extensions:

* By default, there is a new option called "autofallthrough" in
  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier)
  behavior was to wait for an extension to be dialed after there were no
  more extensions to execute.  "autofallthrough" changes this behavior
  so that the call will immediately be terminated with BUSY,
  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
  writing an extension for IVR, you must use the WaitExten application
  if "autofallthrough" is set to yes.

AGI:

* AGI scripts did not always get SIGHUP at the end, previously.  That
  behavior has been fixed.  If you do not want your script to terminate
  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
  be ignored within your application.

* CallerID is reported with agi_callerid and agi_calleridname instead
  of a single parameter holding both.

Music On Hold:

* The preferred format for musiconhold.conf has changed; please see the
  sample configuration file for the new format. The existing format
  is still supported but will generate warnings when the module is loaded.

chan_modem:

* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
  in this release, and will be removed in the next major Asterisk release.
  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
  path for aopen and bestdata modem users.

MeetMe:

* The conference application now allows users to increase/decrease their
  speaking volume and listening volume (independently of each other and
  other users); the 'admin' and 'user' menus have changed, and new sound
  files are included with this release. However, if a user calling in
  over a Zaptel channel that does NOT have hardware DTMF detection
  increases their speaking volume, it is likely they will no longer be
  able to enter/exit the menu or make any further adjustments, as the
  software DTMF detector will not be able to recognize the DTMF coming
  from their device.

GetVar Manager Action:

* Previously, the behavior of the GetVar manager action reported the value
  of a variable in the following manner:
   > name: value
  This has been changed to a manner similar to the SetVar action and is now
   > Variable: name
   > Value: value