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
This commit is contained in:
Matthew Jordan
parent
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362
CHANGES
362
CHANGES
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@ -8,79 +8,142 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
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------------------------------------------------------------------------------
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accountcode
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Overview
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------------------
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- Added functional peeraccount support. Except for Queue, the
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accountcode propagation is now consistently propagated to outgoing
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channels before dialing. The channel accountcode can change from its
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original non-empty value on channel creation for the following specific
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reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
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originate method that can specify an accountcode value. Three, the
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calling channel propagates its peeraccount or accountcode to the
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outgoing channel's accountcode before dialing. The change has two
|
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visible effects. One, local channels now cross accountcode and
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peeraccount across the special bridge between the ;1 and ;2 channels
|
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just like channels between normal bridges. Two, the
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CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
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set the accountcode on the outgoing channel(s).
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For Queue, an outgoing channel's non-empty accountcode will not change
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unless explicitly set by CHANNEL(accountcode). The change has three
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visible effects. One, local channels now cross accountcode and
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peeraccount across the special bridge between the ;1 and ;2 channels
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just like channels between normal bridges. Two, the queue member will
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get an accountcode if it doesn't have one and one is available from the
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calling channel's peeraccount. Three, accountcode propagation includes
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local channel members where the accountcodes are propagated early
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enough to be available on the ;2 channel.
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Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
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the focus of development for this release of Asterisk was on improving the
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usability and features developed in the previous Standard release, Asterisk 12.
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Beyond a general refinement of end user features, development focussed heavily
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on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
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REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
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new features include:
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app_dahdibarge
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* Asterisk security events are now provided via AMI, allowing end users to
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monitor their Asterisk system in real time for security related issues.
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* External control of Message Waiting Indicators (MWI) through both AMI and ARI.
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* Reception/transmission of out of call text messages using any supported
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channel driver/protocol stack through ARI.
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* Resource List Server support in the PJSIP stack, providing subscriptions to
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lists of resources and batched delivery of NOTIFY requests.
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* Inter-Asterisk distributed device state and mailbox state using the PJSIP
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stack.
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It is important to note that Asterisk 13 is built on the architecture developed
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during the previous Standard release, Asterisk 12. Users upgrading to
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Asterisk 13 should read about the new features in Asterisk 12 later in this file
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(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
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UPGRADE-12.txt delivered with this release. In particular, users upgrading to
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Asterisk 13 from a release prior to Asterisk 12 should read the specifications
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on AMI, CDRs, and CEL on the Asterisk wiki:
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* AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
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* CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
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* CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
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Many new featuers in Asterisk 13 were introduced in point releases of
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Asterisk 12. Following this section - which documents the changes from all
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versions of Asterisk 12 to Asterisk 13 - users should examine the new features
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that were introduced in the point releases of Asterisk 12, as they are also
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included in Asterisk 13.
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Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
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delivered with this release.
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Build System
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------------------
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* Sample config files have been moved from configs/ to a sub-folder of that
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directory, samples.
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* The menuselect utility has been pulled into the Asterisk repository. As a
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result, the libxml2 development library is now a required dependency for
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Asterisk.
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* A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
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counted objects will emit additional debug information to the refs log file
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located in the standard Asterisk log file directory. This log file is useful
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in tracking down object leaks and other reference counting issues. Prior to
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this version, this option was only available by modifying the source code
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directly. This change also includes a new script, refcounter.py, in the
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contrib folder that will process the refs log file. Note that this replaces
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the refcounter utility that could be built from the utils directory.
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Applications
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------------------
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DahdiBarge
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------------------
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* This module was deprecated and has been removed. Users of app_dahdibarge
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should use ChanSpy instead.
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app_readfile
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MixMonitor
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------------------
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* New options to play a beep when starting a recording and stopping a recording
|
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have been added. The option "p" will play a beep to the channel that starts
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the recording. The option "P" will play a beep to the channel that stops the
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recording.
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ReadFile
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------------------
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* This module was deprecated and has been removed. Users of app_readfile
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should use func_env's FILE function instead.
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app_saycountpl
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Say
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------------------
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* The 'say' family of dialplan applications now support the Japanese
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language. The 'language' parameter in say.conf now recognizes a setting of
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'ja', which will enable Japanese language specific mechanisms for playing
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back numbers, dates, and other items.
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SayCountPL
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------------------
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* This module was deprecated and has been removed. Users of app_saycountpl
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should use the Say family of applications.
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AMI
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SetMusicOnHold
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------------------
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* New DeviceStateChanged and PresenceStateChanged AMI events have been added.
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These events are emitted whenever a device state or presence state change
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occurs. The events are controlled by res_manager_device_state.so and
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res_manager_presence_state.so. If the high frequency of these events is
|
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problematic for you, do not load these modules.
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* The SetMusicOnHold dialplan application was deprecated and has been removed.
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Users of the application should use the CHANNEL function's musicclass
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setting instead.
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* Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
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work in basically the same way as the 'dialplan add extension' and
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'dialplan remove extension' CLI commands respectively.
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WaitMusicOnHold
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------------------
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* The WaitMusicOnHold dialplan application was deprecated and has been
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removed. Users of the application should use MusicOnHold with a duration
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parameter instead.
|
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* New AMI action LoggerRotate reloads and rotates logger in the same manner
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as CLI command 'logger rotate'
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VoiceMail
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||||
------------------
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* VoiceMail and VoiceMailMain now support the Japanese language. The
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'language' parameter in voicemail.conf now recognizes a setting of 'ja',
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which will enable prompts to be played back using a Japanese grammatical
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structure. Additional prompts are necessary for this functionality,
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including:
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- jb-arimasu: there is
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||||
- jb-arimasen: there is not
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- jb-oshitekudasai: please press
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- jb-ni: article ni
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||||
- jb-ga: article ga
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- jb-wa: article wa
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||||
- jb-wo: article wo
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||||
|
||||
* New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
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functionality of CLI commands 'fax show sessions', 'fax show session',
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and fax show stats' respectively.
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* Add the ability to specify multiple email addresses in configuration,
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||||
separated by a |.
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* New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
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||||
enable manager control over PRI debugging levels and file output.
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||||
|
||||
* AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
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endpoint as long as a default outbound endpoint is set. This also applies
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to the equivalent CLI command (pjsip send notify)
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* The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
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that give information on Asterisk's attempts to qualify the endpoint.
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CDR Backends
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------------------
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cdr_sqlite
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-----------------
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@ -94,6 +157,10 @@ cdr_pgsql
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pg_stat_activity view and CSV log entries. This setting is configurable
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for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
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|
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CEL Backends
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------------------
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cel_pgsql
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------------------
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||||
* Added the ability to support PostgreSQL application_name on connections.
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@ -101,10 +168,9 @@ cel_pgsql
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pg_stat_activity view and CSV log entries. This setting is configurable
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for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
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||||
CEL
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Channel Drivers
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------------------
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* The "bridge_technology" extra field key has been added to BRIDGE_ENTER
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and BRIDGE_EXIT events.
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chan_dahdi
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------------------
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@ -142,7 +208,76 @@ chan_sip
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* The SIPCHANINFO dialplan function was deprecated and has been removed. Users
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of the function should use the CHANNEL function instead.
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Core
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||||
------------------
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||||
|
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Account Codes
|
||||
------------------
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||||
* 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
|
||||
|
|
@ -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
|
||||
|
||||
Resources
|
||||
------------------
|
||||
* 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
|
||||
------------------
|
||||
* 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 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.
|
||||
|
||||
* 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.
|
||||
|
||||
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.
|
||||
|
|
|
|||
Loading…
Reference in New Issue