Remove unneeded CHANGES and UPGRADE files

(cherry picked from commit ea7e719d71)
This commit is contained in:
George Joseph 2023-08-09 11:08:20 -06:00 committed by Asterisk Development Team
parent 68349125a8
commit f7ea380b04
52 changed files with 0 additions and 363 deletions

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Subject: app_senddtmf
The SendFlash AMI action now allows sending
a hook flash event on a channel.

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Subject: pbx_builtins
It is now possible to not wait for media on
a channel when answering it using Answer,
by specifying the i option.

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Subject: app_amd
An audio file to play during AMD processing can
now be specified to the AMD application or configured
in the amd.conf configuration file.

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Subject: app_bridgewait
Adds the n option to not answer the channel when
the BridgeWait application is called.

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Subject: app_broadcast
A Broadcast application is now available which allows
for asynchronous one-to-many and many-to-one channel audio.

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Subject: app_confbridge
Adds the end_marked_any option which can be used
to kick users from a conference after any
marked user leaves (including marked users).

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Subject: app_directory
A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.

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Subject: app_if
Adds the If, ElseIf, Else, EndIf, and ExitIf applications
for conditional execution of a block of code.

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Subject: app_mixmonitor
Adds the c option to use the real Caller ID on
the channel in voicemail recordings as opposed
to the Connected Line.

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Subject: app_mixmonitor
The d option for MixMonitor now allows deleting
the original recording when MixMonitor exits,
which can be useful when MixMonitor copies it
somewhere else before exiting.

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Subject: app_mixmonitor
Subject: audiohook
Subject: manager
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.

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Subject: app_read
A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.

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Subject: app_senddtmf
A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.

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Subject: app_signal
Adds Signal and WaitForSignal applications
which can be used for signaling or as a
simple message queue in the dialplan.

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Subject: app_voicemail
The voicemail user option attachextrecs can
now be set to control whether external recordings
trigger voicemail email notifications.

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Subject: bridge_builtin_features
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.

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Subject: cdr
Two new options have been added which allow
bridging and dial state changes to be ignored
in CDRs, which can be useful if a single CDR
is desired for a channel.

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Subject: cli
Subject: core
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

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Subject: db
The DBPrefixGet AMI action now allows retrieving
all of the DB keys beginning with a particular
prefix.

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Subject: DUNDi
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.

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Subject: features
The Bridge application now has the n "no answer" option
that can be used to prevent the channel from being
automatically answered prior to bridging.

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Subject: format_sln
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.

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@ -1,5 +0,0 @@
Subject: New EXPORT function
A new function, EXPORT, allows writing variables
and functions on other channels, the complement
of the IMPORT function.

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Subject: func_json
Additional parsing capabilities have been added to the
JSON_DECODE function, including support for arrays
and recursive indexing.

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Subject: func_strings
Three new functions, TRIM, LTRIM, and RTRIM, are
now available for trimming leading and trailing
whitespace.

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Subject: chan_dahdi
FXO channels (FXS signaled) that don't use callerid or
distinctive ring detection can now be configured
to enter the dialplan immediately using immediate=yes,
instead of waiting for at least one ring.

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Subject: locks
A new AMI event, DeadlockStart, is now available
when Asterisk is compiled with DETECT_DEADLOCKS,
and can indicate that a deadlock has occured.

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Subject: AMI
The AOCMessage action can now be used to generate AOC-S messages.

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Subject: res_geolocation
* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
Added 4 built-in profiles:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
The profiles are empty except for having their precedence
set.
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles. This is
mutually exclusive with setting location_reference on the
profile.
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.
Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.

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Subject: Add support for named capture agent.
A name for the capture agent can now be specified
using the capture_name option which, if specified,
will be sent to the HEP server.

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Subject: res_http_media_cache
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs

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Subject: res_musiconhold_answeredonly
This change adds an option, answeredonly, that will prevent music
on hold on channels that are not answered.

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Subject: res_phoneprov
On multihomed Asterisk servers with dynamic SERVER template variables,
reloading this module is no longer required when re-provisioning your
phone to another interface address (e.g. when moving between VLANs.)

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Subject: res_pjsip
A new option named "peer_supported" has been added to the endpoint option
100rel. When set to this option, Asterisk sends provisional responses
reliably if the peer supports it. If the peer does not support reliable
provisional responses, Asterisk sends them normally.

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Subject: res_pjsip
A new option named "all_codecs_on_empty_reinvite" has been added to the
global section. When this option is enabled, on reception of a re-INVITE
without SDP, Asterisk will send an SDP offer in the 200 OK response containing
all configured codecs on the endpoint, instead of simply those that have
already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
The default value is "off".

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Subject: res_pjsip_aoc
Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
A new endpoint option, send_aoc, controls this.

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Subject: res_pjsip_logger
SIP messages can now be filtered by SIP request method
(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
allowing for more granular debugging to be done
in the CLI. This applies to requests but not responses.

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Subject: res_pjsip_notify
Allows using the config options in pjsip_notify.conf
from AMI actions as with the existing CLI commands.

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Subject: res_pjsip_header_funcs
The new PJSIP_HEADER_PARAM function now fully supports both
URI and header parameters. Both reading and writing
parameters are supported.

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Subject: res_pjsip
Added options "security_negotiation" and "security_mechanisms" to pjsip
endpoints and registrations. "security_negotiation" can be set to "no" (default)
or "mediasec", and "security_mechanisms" can be a list of comma-separated
security_mechanisms in the form defined by RFC 3329 section 2.2.

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Subject: res_pjsip_session
The overlap_context option now allows explicitly
specifying a context to use for overlap dialing matches.

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Subject: res_pjsip
TLS transports in res_pjsip can now reload their TLS certificate
and private key files, provided the filename of them has not
changed.

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subject: res_pjsip
user_eq_phone=yes flag on a pjsip endpoint will now set user=phone on
the From and Prviacy headers in addition to the existing To and RURI

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Subject: res_rtp_asterisk
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is available using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

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Subject: res_pjsip_rfc3326
Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
addition to currently supported Q.850). The first header found will be used to set
the HANGUPCAUSE variable.

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Subject: res_tonedetect
The TONE_DETECT function now supports
detection of audible ringback tone
using the p option.

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Subject: test.c
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.

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Subject: Transfer feature
The following capabilities have been added to the
transfer feature:
- The transfer initiation announcement prompt can
now be customized in features.conf.
- The TRANSFER_EXTEN variable now can be set on the
transferer's channel in order to allow the transfer
function to automatically attempt to go to the extension
contained in this variable, if it exists. The transfer
context behavior is not changed (TRANSFER_CONTEXT is used
if it exists; otherwise the default context is used).

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Subject: xmldocs
The XML documentation can now be reloaded without restarting
Asterisk, which makes it possible to load new modules that
enforce documentation without restarting Asterisk.

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Subject: app_playback
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.

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Subject: AMI (Asterisk Manager Interface)
Previously, GetConfig and UpdateConfig were able to access files outside of
the Asterisk configuration directory. Now this access is put behind the
live_dangerously configuration option in asterisk.conf, which is disabled by
default. If access to configuration files outside of the Asterisk configuation
directory is required via AMI, then the live_dangerously configuration option
must be set to yes.

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Subject: res_crypto
In addition to only paying attention to files ending with .key or .pub
in the keys directory, we now also ignore any files which aren't regular
files.