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=========================================================
=== Information for upgrading from Asterisk 1.4 to 1.6
===
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt     -- Upgrade info for 1.4 to 1.6
=========================================================

AEL:

* Macros are now implemented underneath with the Gosub() application.
  Heaven Help You if you wrote code depending on any aspect of this!
  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.

* 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.

* The following core commands dealing with dialplan has been deprecated: 'core
  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
  instead.

* The silencethreshold used for various applications is now settable via a
  centralized config option in dsp.conf.

* The logical value of spaces immediately preceding a standalone 0 previously
  evaluated to true.  It now evaluates to false.  This has confused a good
  many people in the past (typically because they failed to realize the space
  had any significance).  Since this violates the Principle of Least Surprise,
  it has been changed.

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.

* The externnotify script should accept an additional (last) parameter
  containing the number of urgent messages in the INBOX.

Applications:

* SendImage() no longer hangs up the channel on transmission error or on
  another type of error; in those cases, a FAILURE status is stored in 
  SENDIMAGESTATUS and dialplan execution continues.  The possible return values
  stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and UNSUPPORTED. ('OK' has
  been replaced with 'SUCCESS', and 'NOSUPPORT' has been replaced with
  'UNSUPPORTED').  This change makes the SendImage application more consistent
  with other 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.

* While app_directory has always relied on having a voicemail.conf or users.conf file
  correctly set up, it now is dependent on app_voicemail being compiled as well.

* 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: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
  and you should start using that function instead for retrieving information about
  the channel in a technology-agnostic way.

* 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.

* If you have any third party modules which use a config file variable whose
  name ends in a '+', please note that the append capability added to this
  version may now conflict with that variable naming scheme.  An easy
  workaround is to ensure that a space occurs between the '+' and the '=',
  to differentiate your variable from the append operator.  This potential
  conflict is unlikely, but is documented here to be thorough.

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.

Queues:

* New queue log events ADDMEMBER and REMOVEMEMBER have been added.  Also, a
  new value has been added to the TRANSFER event that indicates the caller's
  original position in the queue they are being transfered from.

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.