Shuffle UPGRADE.txt files for 1.10.

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279118 f38db490-d61c-443f-a65b-d21fe96a405b

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+===========================================================
+===
+=== Information for upgrading between Asterisk 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 to 1.8:
+
+* The behavior of the 'parkedcallstimeout' has changed slightly.  The formulation
+  of the extension name that a timed out parked call is delivered to when this
+  option is set to 'no' was modified such that instead of converting '/' to '0',
+  the '/' is converted to an underscore '_'.  See the updated documentation in
+  features.conf.sample for more information on the behavior of the
+  'parkedcallstimeout' option.
+
+* 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. 
+
+* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
+  1.6 behavior by default, if there is no [compat] section in asterisk.conf.  In
+  prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
+  Specifically, that means that pbx_realtime and res_agi expect you to use commas
+  to separate arguments in applications, and Set only takes a single pair of
+  a variable name/value.  The old 1.4 behavior may still be obtained by setting
+  app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
+  asterisk.conf.
+
+* The PRI channels in chan_dahdi can no longer change the channel name if a
+  different B channel is selected during call negotiation.  To prevent using
+  the channel name to infer what B channel a call is using and to avoid name
+  collisions, the channel name format is changed.
+  The new channel naming for PRI channels is:
+  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
+
+* The ChanIsAvail application has been changed so the AVAILSTATUS variable
+  no longer contains both the device state and cause code. The cause code
+  is now available in the AVAILCAUSECODE variable. If existing dialplan logic
+  is written to expect AVAILSTATUS to contain the cause code it needs to be
+  changed to use AVAILCAUSECODE.
+
+* ExternalIVR will now send Z events for invalid or missing files, T events
+  now include the interrupted file and bugs in argument parsing have been
+  fixed so there may be arguments specified in incorrect ways that were
+  working that will no longer work.
+  Please see doc/externalivr.txt for details.
+
+* OSP lookup application changes following variable names:
+  OSPPEERIP to OSPINPEERIP
+  OSPTECH to OSPOUTTECH
+  OSPDEST to OSPDESTINATION
+  OSPCALLING to OSPOUTCALLING
+  OSPCALLED to OSPOUTCALLED
+  OSPRESULTS to OSPDESTREMAILS
+
+* The Manager event 'iax2 show peers' output has been updated.  It now has a
+  similar output of 'sip show peers'.
+
+* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
+  of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
+  the current dialplan context.
+
+* The CALLERPRES() dialplan function is deprecated in favor of
+  CALLERID(num-pres) and CALLERID(name-pres).
+
+* Environment variables that start with "AST_" are reserved to the system and
+  may no longer be set from the dialplan.
+
+* When a call is redirected inside of a Dial, the app and appdata fields of the
+  CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
+
+* The CDR handling of billsec and duration field has changed. If your table
+  definition specifies those fields as float,double or similar they will now
+  be logged with microsecond accuracy instead of a whole integer.
+
+* chan_sip will no longer set up a local call forward when receiving a
+  482 Loop Detected response. The dialplan will just continue from where it
+  left off.
+
+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.
+
+* 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".
+
+* 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.
+
+* The cdr.conf file must exist and be configured correctly in order for CDR
+  records to be written.
+
+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.
+