File: UPGRADE.txt

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===========================================================
===
=== 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
===
===========================================================

from 1.6.2.9-2+squeeze11 to 1.6.2.9-2+squeeze12 (backported from 1.8.24.1):
* Certain dialplan functions have been marked as 'dangerous', and may only be
  executed from the dialplan. Execution from extenal sources (AMI's GetVar and
  SetVar actions; etc.) may be inhibited by setting live_dangerously in the
  [options] section of asterisk.conf to no. SHELL(), channel locking, and direct
  file read/write functions are marked as dangerous. DB_DELETE() and
  REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
  accept writes (which ignore the provided value).

From 1.6.1 to 1.6.2:

* SIP no longer sends the 183 progress message for early media by
  default.  Applications requiring early media should use the
  progress() dialplan app to generate the progress message. 

* The firmware for the IAXy has been removed from Asterisk.  It can be
  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
  install the firmware into its proper location, place the firmware in the
  contrib/firmware/iax/ directory in the Asterisk source tree before running
  "make install".

* T.38 FAX error correction mode can no longer be configured in udptl.conf;
  instead, it is configured on a per-peer (or global) basis in sip.conf, with
  the same default as was present in udptl.conf.sample.

* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
  instead, it is either supplied by the application servicing the T.38 channel
  (for a FAX send or receive) or calculated from the bridged endpoint's
  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
  allows for overriding the value supplied by a remote endpoint, which is useful
  when T.38 connections are made to gateways that supply incorrectly-calculated
  maximum datagram sizes.

* There have been some changes to the IAX2 protocol to address the security
  concerns documented in the security advisory AST-2009-006.  Please see the
  IAX2 security document, doc/IAX2-security.pdf, for information regarding
  backwards compatibility with versions of Asterisk that do not contain these
  changes to IAX2.

* Beginning with this release, Asterisk's internal methods of
  negotiating T.38 (FAX over IP) sessions changed in
  non-backwards-compatible ways. Any applications that previously used
  AST_CONTROL_T38 control frames will have to be upgraded to use
  AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
  example of how to generate and respond to these frames. These changes
  were made to solve significant T.38 interoperability problems between
  Asterisk and various SIP/T.38 endpoints identified by many users of
  Asterisk.

* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
  has been renamed to 'directmedia', to better reflect what it actually does.
  In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
  starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
  option never had any effect on these cases, it only affected the re-INVITEs
  used for direct media path setup. For MGCP and Skinny, the option was poorly
  named because those protocols don't even use INVITE messages at all. For
  backwards compatibility, the old option is still supported in both normal
  and Realtime configuration files, but all of the sample configuration files,
  Realtime/LDAP schemas, and other documentation refer to it using the new name.

* 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.

* 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.

* Support for Taiwanese was incorrectly supported with the "tw" language code.
  In reality, the "tw" language code is reserved for the Twi language, native
  to Ghana.  If you were previously using the "tw" language code, you should
  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
  specific localizations.  Additionally, "mx" should be changed to "es_MX",
  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
  "cs", not "cz".

* The prematuremedia option in sip.conf is from this released enabled by
  default. See sip.conf.sample

* DAHDISendCallreroutingFacility() parameters are now comma-separated,
  instead of the old pipe.

* res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
  that would end up being interpreted as a bug once Asterisk started removing 
  the contacts from a user list.

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.

* If you use ODBC storage for voicemail, there is a new field called "flag"
  which should be a char(8) or larger.  This field specifies whether or not a
  message has been designated to be "Urgent", "PRIORITY", or not.