UPGRADE.txt

===========================================================
===
=== Information for upgrading between Asterisk 1.6 versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== 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-1.6.txt -- Upgrade info for 1.4 to 1.6
===
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From 1.6.2 to 1.6.3:

* Asterisk-addons no longer exists as an independent package.  Those modules
  now live in the addons directory of the main Asterisk source tree.  They
  are not enabled by default.  For more information about why modules live in
  addons, see README-addons.txt.

* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
  users of this channel in the tree have been converted to LOG_NOTICE or removed
  (in cases where the same message was already generated to another channel).

* The usage of RTP inside of Asterisk has now become modularized. This means
  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
  If you are not using autoload=yes in modules.conf you will need to ensure
  it is set to load. If not, then any module which uses RTP (such as chan_sip)
  will not be able to send or receive calls.

* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still 
  remains. It now exists within app_chanspy.c and retains the exact same 
  functionality as before. 

From 1.6.1 to 1.6.2:

* The res_indications module has been removed.  Its functionality was important
  enough that most of it has been moved into the Asterisk core.
  Two applications previously provided by res_indications, PlayTones and
  StopPlayTones, have been moved into a new module, app_playtones.

From 1.6.0.1 to 1.6.1:

* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
  API calls were added in 1.6.0, so that modules that provide multiple
  AGI commands could register/unregister them all with a single
  step. However, these API calls were not implemented properly, and did
  not allow the caller to know whether registration or unregistration
  succeeded or failed. They have been redefined to now return success
  or failure, but this means any code using these functions will need
  be recompiled after upgrading to a version of Asterisk containing
  these changes. In addition, the source code using these functions
  should be reviewed to ensure it can properly react to failure
  of registration or unregistration of its API commands.

* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
  to better match what it really does, and the argument order has been
  changed to be consistent with other API calls that perform similar
  operations.

From 1.6.0.x to 1.6.1:

* In previous versions of Asterisk, due to the way objects were arranged in
  memory by chan_sip, the order of entries in sip.conf could be adjusted to
  control the behavior of matching against peers and users.  The way objects
  are managed has been significantly changed for reasons involving performance
  and stability.  A side effect of these changes is that the order of entries
  in sip.conf can no longer be relied upon to control behavior.

* The following core commands dealing with dialplan have 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.

* In the dialplan expression parser, 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.

* 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.

* 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.

* 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.

* The "Join" event from app_queue now uses the CallerIDNum header instead of
  the CallerID header to indicate the CallerID number.

From 1.6.1 to 1.6.2:

* The default console now will use colors according to the default background
  color, instead of forcing the background color to black.  If you are using a
  light colored background for your console, you may wish to use the option
  flag '-W' to present better color choices for the various messages.  However,
  if you'd prefer the old method of forcing colors to white text on a black
  background, the compatibility option -B is provided for this purpose.

* SendImage() no longer hangs up the channel on transmission error or on
  any other 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.

* skinny.conf now has separate sections for lines and devices.
  Please have a look at configs/skinny.conf.sample and update
  your skinny.conf.

* Queue names previously were treated in a case-sensitive manner,
  meaning that queues with names like "sales" and "sALeS" would be
  seen as unique queues. The parsing logic has changed to use
  case-insensitive comparisons now when originally hashing based on
  queue names, meaning that now the two queues mentioned as examples
  earlier will be seen as having the same name.

* The SPRINTF() dialplan function has been moved into its own module,
  func_sprintf, and is no longer included in func_strings. If you use this
  function and do not use 'autoload=yes' in modules.conf, you will need
  to explicitly load func_sprintf for it to be available.