Update CHANGES file

........

Merged revisions 420609 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 241 insertions, 115 deletions

diff --git a/CHANGES b/CHANGES
index 640945cd0..dc88d1f38 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,78 +9,141 @@
 ==============================================================================
 
 ------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
+------------------------------------------------------------------------------
+
+
+
+------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
 ------------------------------------------------------------------------------
 
-accountcode
+Overview
 ------------------
- - Added functional peeraccount support.  Except for Queue, the
-   accountcode propagation is now consistently propagated to outgoing
-   channels before dialing.  The channel accountcode can change from its
-   original non-empty value on channel creation for the following specific
-   reasons.  One, dialplan sets it using CHANNEL(accountcode).  Two, an
-   originate method that can specify an accountcode value.  Three, the
-   calling channel propagates its peeraccount or accountcode to the
-   outgoing channel's accountcode before dialing.  The change has two
-   visible effects.  One, local channels now cross accountcode and
-   peeraccount across the special bridge between the ;1 and ;2 channels
-   just like channels between normal bridges.  Two, the
-   CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
-   set the accountcode on the outgoing channel(s).
 
-   For Queue, an outgoing channel's non-empty accountcode will not change
-   unless explicitly set by CHANNEL(accountcode).  The change has three
-   visible effects.  One, local channels now cross accountcode and
-   peeraccount across the special bridge between the ;1 and ;2 channels
-   just like channels between normal bridges.  Two, the queue member will
-   get an accountcode if it doesn't have one and one is available from the
-   calling channel's peeraccount.  Three, accountcode propagation includes
-   local channel members where the accountcodes are propagated early
-   enough to be available on the ;2 channel.
+Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
+the focus of development for this release of Asterisk was on improving the
+usability and features developed in the previous Standard release, Asterisk 12.
+Beyond a general refinement of end user features, development focussed heavily
+on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
+REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
+new features include:
+
+* Asterisk security events are now provided via AMI, allowing end users to
+  monitor their Asterisk system in real time for security related issues.
+* External control of Message Waiting Indicators (MWI) through both AMI and ARI.
+* Reception/transmission of out of call text messages using any supported
+  channel driver/protocol stack through ARI.
+* Resource List Server support in the PJSIP stack, providing subscriptions to
+  lists of resources and batched delivery of NOTIFY requests.
+* Inter-Asterisk distributed device state and mailbox state using the PJSIP
+  stack.
+
+It is important to note that Asterisk 13 is built on the architecture developed
+during the previous Standard release, Asterisk 12. Users upgrading to
+Asterisk 13 should read about the new features in Asterisk 12 later in this file
+(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
+UPGRADE-12.txt delivered with this release. In particular, users upgrading to
+Asterisk 13 from a release prior to Asterisk 12 should read the specifications
+on AMI, CDRs, and CEL on the Asterisk wiki:
+ * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
+ * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
+ * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
 
-app_dahdibarge
+Many new featuers in Asterisk 13 were introduced in point releases of
+Asterisk 12. Following this section - which documents the changes from all
+versions of Asterisk 12 to Asterisk 13 - users should examine the new features
+that were introduced in the point releases of Asterisk 12, as they are also
+included in Asterisk 13.
+
+Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
+delivered with this release.
+
+
+Build System
+------------------
+ * Sample config files have been moved from configs/ to a sub-folder of that
+   directory, samples.
+
+ * The menuselect utility has been pulled into the Asterisk repository. As a
+   result, the libxml2 development library is now a required dependency for
+   Asterisk.
+
+ * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
+   counted objects will emit additional debug information to the refs log file
+   located in the standard Asterisk log file directory. This log file is useful
+   in tracking down object leaks and other reference counting issues. Prior to
+   this version, this option was only available by modifying the source code
+   directly. This change also includes a new script, refcounter.py, in the
+   contrib folder that will process the refs log file. Note that this replaces
+   the refcounter utility that could be built from the utils directory.
+
+
+Applications
+------------------
+
+DahdiBarge
 ------------------
  * This module was deprecated and has been removed. Users of app_dahdibarge
    should use ChanSpy instead.
 
-app_readfile
+MixMonitor
+------------------
+ * New options to play a beep when starting a recording and stopping a recording
+   have been added.  The option "p" will play a beep to the channel that starts
+   the recording.  The option "P" will play a beep to the channel that stops the
+   recording.
+
+ReadFile
 ------------------
  * This module was deprecated and has been removed. Users of app_readfile
    should use func_env's FILE function instead.
 
-app_saycountpl
+Say
+------------------
+ * The 'say' family of dialplan applications now support the Japanese
+   language. The 'language' parameter in say.conf now recognizes a setting of
+   'ja', which will enable Japanese language specific mechanisms for playing
+   back numbers, dates, and other items.
+
+SayCountPL
 ------------------
  * This module was deprecated and has been removed. Users of app_saycountpl
    should use the Say family of applications.
 
-AMI
+SetMusicOnHold
 ------------------
- * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
-   These events are emitted whenever a device state or presence state change
-   occurs. The events are controlled by res_manager_device_state.so and
-   res_manager_presence_state.so. If the high frequency of these events is
-   problematic for you, do not load these modules.
+ * The SetMusicOnHold dialplan application was deprecated and has been removed.
+   Users of the application should use the CHANNEL function's musicclass
+   setting instead.
 
- * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
-   work in basically the same way as the 'dialplan add extension' and
-   'dialplan remove extension' CLI commands respectively.
+WaitMusicOnHold
+------------------
+ * The WaitMusicOnHold dialplan application was deprecated and has been
+   removed. Users of the application should use MusicOnHold with a duration
+   parameter instead.
 
- * New AMI action LoggerRotate reloads and rotates logger in the same manner
-   as CLI command 'logger rotate'
+VoiceMail
+------------------
+ * VoiceMail and VoiceMailMain now support the Japanese language. The
+   'language' parameter in voicemail.conf now recognizes a setting of 'ja',
+   which will enable prompts to be played back using a Japanese grammatical
+   structure. Additional prompts are necessary for this functionality,
+   including:
+   - jb-arimasu: there is
+   - jb-arimasen: there is not
+   - jb-oshitekudasai: please press
+   - jb-ni: article ni
+   - jb-ga: article ga
+   - jb-wa: article wa
+   - jb-wo: article wo
 
- * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
-   functionality of CLI commands 'fax show sessions', 'fax show session',
-   and fax show stats' respectively.
+ * Add the ability to specify multiple email addresses in configuration,
+   separated by a |.
 
- * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
-   enable manager control over PRI debugging levels and file output.
 
- * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
-   endpoint as long as a default outbound endpoint is set. This also applies
-   to the equivalent CLI command (pjsip send notify)
- 
- * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
-   that give information on Asterisk's attempts to qualify the endpoint.
+CDR Backends
+------------------
 
 cdr_sqlite
 -----------------
@@ -94,6 +157,10 @@ cdr_pgsql
    pg_stat_activity view and CSV log entries. This setting is configurable
    for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
 
+
+CEL Backends
+------------------
+
 cel_pgsql
 ------------------
  * Added the ability to support PostgreSQL application_name on connections.
@@ -101,10 +168,9 @@ cel_pgsql
    pg_stat_activity view and CSV log entries. This setting is configurable
    for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
 
-CEL
+
+Channel Drivers
 ------------------
- * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
-   and BRIDGE_EXIT events.
 
 chan_dahdi
 ------------------
@@ -142,8 +208,77 @@ chan_sip
  * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
    of the function should use the CHANNEL function instead.
 
+
 Core
 ------------------
+
+Account Codes
+------------------
+ * Added functional peeraccount support.  Except for Queue, the
+   accountcode propagation is now consistently propagated to outgoing
+   channels before dialing.  The channel accountcode can change from its
+   original non-empty value on channel creation for the following specific
+   reasons.  One, dialplan sets it using CHANNEL(accountcode).  Two, an
+   originate method that can specify an accountcode value.  Three, the
+   calling channel propagates its peeraccount or accountcode to the
+   outgoing channel's accountcode before dialing.  The change has two
+   visible effects.  One, local channels now cross accountcode and
+   peeraccount across the special bridge between the ;1 and ;2 channels
+   just like channels between normal bridges.  Two, the
+   CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
+   set the accountcode on the outgoing channel(s).
+
+   For Queue, an outgoing channel's non-empty accountcode will not change
+   unless explicitly set by CHANNEL(accountcode).  The change has three
+   visible effects.  One, local channels now cross accountcode and
+   peeraccount across the special bridge between the ;1 and ;2 channels
+   just like channels between normal bridges.  Two, the queue member will
+   get an accountcode if it doesn't have one and one is available from the
+   calling channel's peeraccount.  Three, accountcode propagation includes
+   local channel members where the accountcodes are propagated early
+   enough to be available on the ;2 channel.
+
+AMI
+------------------
+ * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
+   These events are emitted whenever a device state or presence state change
+   occurs. The events are controlled by res_manager_device_state.so and
+   res_manager_presence_state.so. If the high frequency of these events is
+   problematic for you, do not load these modules.
+
+ * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
+   work in basically the same way as the 'dialplan add extension' and
+   'dialplan remove extension' CLI commands respectively.
+
+ * New AMI action LoggerRotate reloads and rotates logger in the same manner
+   as CLI command 'logger rotate'
+
+ * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
+   functionality of CLI commands 'fax show sessions', 'fax show session',
+   and fax show stats' respectively.
+
+ * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
+   enable manager control over PRI debugging levels and file output.
+
+ * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
+   endpoint as long as a default outbound endpoint is set. This also applies
+   to the equivalent CLI command (pjsip send notify)
+ 
+ * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
+   that give information on Asterisk's attempts to qualify the endpoint.
+
+CEL
+------------------
+ * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
+   and BRIDGE_EXIT events.
+
+Features
+------------------
+ * Channel variables are now substituted in arguments passed to applications
+   run by using dynamic features.
+
+TLS
+------------------
  * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
    Enabling PFS is attempted by default, and is dependent on the configuration
    of the module using TLS.
@@ -162,56 +297,18 @@ Core
      will use PFS when offered by the client. Clients which do not offer PFS
      fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
 
-Features
-------------------
- * The ast_channel_feature_hooks* functions have been added to allow features
-   such as DTMF hooks, interval hooks, and bridge event hooks to be made
-   available to a channel when the channel is bridged. Previously, these
-   features were provided exclusively by the caller of ast_bridge_join()
-   outside of "basic" type bridges.
 
- * Channel variables are now substituted in arguments passed to applications
-   run by using dynamic features.
+Functions
+------------------
 
 JACK_HOOK
 ------------------
  * The JACK_HOOK function now supports audio with a sample rate higher than
    8kHz.
 
-MusicOnHold
-------------------
- * The SetMusicOnHold dialplan application was deprecated and has been removed.
-   Users of the application should use the CHANNEL function's musicclass
-   setting instead.
-
- * The WaitMusicOnHold dialplan application was deprecated and has been
-   removed. Users of the application should use MusicOnHold with a duration
-   parameter instead.
 
-Say
+Resources
 ------------------
- * The 'say' family of dialplan applications now support the Japanese
-   language. The 'language' parameter in say.conf now recognizes a setting of
-   'ja', which will enable Japanese language specific mechanisms for playing
-   back numbers, dates, and other items.
-
-VoiceMail
-------------------
- * VoiceMail and VoiceMailMain now support the Japanese language. The
-   'language' parameter in voicemail.conf now recognizes a setting of 'ja',
-   which will enable prompts to be played back using a Japanese grammatical
-   structure. Additional prompts are necessary for this functionality,
-   including:
-   - jb-arimasu: there is
-   - jb-arimasen: there is not
-   - jb-oshitekudasai: please press
-   - jb-ni: article ni
-   - jb-ga: article ga
-   - jb-wa: article wa
-   - jb-wo: article wo
-
- * Add the ability to specify multiple email addresses in configuration,
-   separated by a |.
 
 res_config_pgsql
 ------------------
@@ -221,12 +318,32 @@ res_config_pgsql
    for res_config_pgsql via the dbappname configuration setting in
    res_pgsql.conf.
 
-MixMonitor
+res_pjsip_outbound_publish
 ------------------
- * New options to play a beep when starting a recording and stopping a recording
-   have been added.  The option "p" will play a beep to the channel that starts
-   the recording.  The option "P" will play a beep to the channel that stops the
-   recording.
+ * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
+   PUBLISH requests for specific event packages to another SIP User Agent.
+
+res_pjsip_pubsub
+------------------
+ * The publish/subscribe core module has been updated to support RFC 4662
+   Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
+   Resource lists are configured in pjsip.conf under a new object type,
+   resource_list. Resource lists can contain either message-summary or presence
+   events, and can be composed of specific resources that provide the event or
+   other resource lists.
+
+ * Inbound publication support is provided by a new object, inbound-publication.
+   This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
+   resource. Which events are accepted is constructed dynamically; see
+   res_pjsip_publish_asterisk for more information.
+
+res_pjsip_publish_asterisk
+------------------
+ * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
+   Asterisk information to other Asterisk servers. This module is intended only
+   for Asterisk to Asterisk exchanges of information. Currently, this includes
+   both mailbox state and device state information.
+
 
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
@@ -362,18 +479,10 @@ res_parking
 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
 ------------------------------------------------------------------------------
 
-Applications
---------------------------
+Record
+------------------
  * Record application now has an option 'o' which allows 0 to act as an exit
    key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
- * Monitor() - A new option, B(), has been added that will turn on a periodic
-   beep while the call is being recorded.
-
-Functions
---------------------------
- * A new function was added: PERIODIC_HOOK.  This allows running a periodic
-   dialplan hook on a channel.  Any audio generated by this hook will be
-   injected into the call.
 
 ChanSpy
 --------------------------
@@ -410,6 +519,11 @@ Directory
      USEREXIT    user pressed '#' from the selection prompt to exit
      FAILED      directory failed in a way that wasn't accounted for. Dang.
 
+Monitor
+------------------
+ * Monitor() - A new option, B(), has been added that will turn on a periodic
+   beep while the call is being recorded.
+
 MusicOnHold
 --------------------------
  * MusicOnHold streams (all modes other than "files") now support wide band
@@ -437,9 +551,11 @@ MixMonitor
 -------------------------
  * A new function, MIXMONITOR, has been added to allow access to individual
    instances of MixMonitor on a channel.
+
  * A new option, B(), has been added that will turn on a periodic beep while the
    call is being recorded.
 
+
 Channel Drivers
 -------------------------
 
@@ -450,20 +566,19 @@ chan_sip
    the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
    the inbound channel.
 
-Debugging
--------------------------
+Core
+------------------
+ * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
+   the new AST_SORCERY diaplan function.
+
  * Core Show Locks output now includes Thread/LWP ID if the platform
    supports this feature.
+
  * New "logger add channel" and "logger remove channel" CLI commands have
    been added to allow creation and deletion of dynamic logger channels
    without configuration changes. These dynamic logger channels will only
    exist until the next restart of asterisk.
 
-Core
-------------------
- * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
-   the new AST_SORCERY diaplan function.
-
 ARI
 ------------------
  * The live recording object on recording events now contains a target_uri
@@ -502,6 +617,17 @@ RealTime
  * A new set of Alembic scripts has been added for CDR tables. This will create
    a 'cdr' table with the default schema that Asterisk expects.
 
+
+Functions
+------------------
+ * A new function was added: PERIODIC_HOOK.  This allows running a periodic
+   dialplan hook on a channel.  Any audio generated by this hook will be
+   injected into the call.
+
+
+Resources
+------------------
+
 res_hep
 ------------------
  * A new module, res_hep, has been added, that acts as a generic packet
@@ -2213,7 +2339,7 @@ chan_unistim
    as per the UNISTIM protocol.
 
  * Fixed issues with dialtone not matching indications.conf and mute stopping rx
-   as well as tx. Also fixed issue with call "Timer" displaying as French "Durée"
+   as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
 
  * Added ability to use multiple lines for a single phone.  This allows multiple
    calls to occur on a single phone, using callwaiting and switching between calls.