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Matthew Jordan 11 years ago
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@ -8,79 +8,142 @@
===
==============================================================================
------------------------------------------------------------------------------
--- 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
------------------
@ -142,7 +208,76 @@ 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
@ -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.

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