Prepare master for Asterisk 22
This commit is contained in:
George Joseph
asterisk-org-access-app[bot]
parent
51a7b18038
commit
c3c82441a2
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@ -1,37 +0,0 @@
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## **DO NOT REMOVE THIS FILE!**
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The only files that should be added to this directory are ones that will be
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used by the release script to update the CHANGES file automatically. The only
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time that it is necessary to add something to the CHANGES-staging directory is
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if you are either adding a new feature to Asterisk or adding new functionality
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to an existing feature. The file does not need to have a meaningful name, but
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it probably should. If there are multiple items that need documenting, you can
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add multiple files, each with their own description. If the message is going to
|
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be the same for each subject, then you can add multiple subject headers to one
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file. The "Subject: xxx" line is case sensitive! For example, if you are making
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a change to PJSIP, then you might add the file "res_pjsip_my_cool_feature.txt" to
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this directory, with a short description of what it does. The files must have
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the ".txt" suffix. If you are adding multiple entries, they should be done in
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the same commit to avoid merge conflicts. Here's an example:
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> Subject: res_pjsip
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> Subject: Core
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>
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> Here's a pretty good description of my new feature that explains exactly what
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> it does and how to use it.
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Here's a master-only example:
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> Subject: res_ari
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> Master-Only: True
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>
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> This change will only go into the master branch. The "Master-Only" header
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> will never be in a change not in master.
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Note that the second subject has another header: "Master-Only". Changes that go
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into the master branch and ONLY the master branch are the only ones that should
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have this header. Also, the value can only be "true" or "True". The
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"Master-Only" part of the header IS case-sensitive, however!
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For more information, check out the wiki page:
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https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
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@ -1,4 +0,0 @@
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Subject: app_senddtmf
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The SendFlash AMI action now allows sending
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a hook flash event on a channel.
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@ -1,5 +0,0 @@
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Subject: pbx_builtins
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It is now possible to not wait for media on
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a channel when answering it using Answer,
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by specifying the i option.
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@ -1,5 +0,0 @@
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Subject: app_amd
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An audio file to play during AMD processing can
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now be specified to the AMD application or configured
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in the amd.conf configuration file.
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@ -1,4 +0,0 @@
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Subject: app_bridgewait
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Adds the n option to not answer the channel when
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the BridgeWait application is called.
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@ -1,4 +0,0 @@
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Subject: app_broadcast
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A Broadcast application is now available which allows
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for asynchronous one-to-many and many-to-one channel audio.
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@ -1,5 +0,0 @@
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Subject: app_confbridge
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Adds the end_marked_any option which can be used
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to kick users from a conference after any
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marked user leaves (including marked users).
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@ -1,5 +0,0 @@
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Subject: app_directory
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A new option 's' has been added to the Directory() application that
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will skip calling the extension and instead set the extension as
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DIRECTORY_EXTEN channel variable.
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@ -1,4 +0,0 @@
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Subject: app_if
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Adds the If, ElseIf, Else, EndIf, and ExitIf applications
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for conditional execution of a block of code.
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Subject: app_mixmonitor
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Adds the c option to use the real Caller ID on
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the channel in voicemail recordings as opposed
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to the Connected Line.
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Subject: app_mixmonitor
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The d option for MixMonitor now allows deleting
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the original recording when MixMonitor exits,
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which can be useful when MixMonitor copies it
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somewhere else before exiting.
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@ -1,17 +0,0 @@
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Subject: app_mixmonitor
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Subject: audiohook
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Subject: manager
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It is now possible to specify the MixMonitorID when calling
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the manager action: MixMonitorMute. This will allow an
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individual MixMonitor instance to be muted via ID.
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The MixMonitorID can be stored as a channel variable using
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the 'i' MixMonitor option and is returned upon creation if
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this option is used.
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As part of this change, if no MixMonitorID is specified in
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the manager action MixMonitorMute, Asterisk will set the mute
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flag on all MixMonitor audiohooks on the channel. Previous
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behavior would set the flag on the first MixMonitor audiohook
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found.
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@ -1,5 +0,0 @@
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Subject: app_read
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A new option 'e' has been added to allow Read() to return the
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terminator as the dialed digits in the case where only the terminator
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is entered.
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@ -1,5 +0,0 @@
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Subject: app_senddtmf
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A new option has been added to SendDTMF() which will answer the
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specified channel if it is not already up. If no channel is specified,
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the current channel will be answered instead.
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@ -1,5 +0,0 @@
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Subject: app_signal
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Adds Signal and WaitForSignal applications
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which can be used for signaling or as a
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simple message queue in the dialplan.
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@ -1,5 +0,0 @@
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Subject: app_voicemail
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The voicemail user option attachextrecs can
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now be set to control whether external recordings
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trigger voicemail email notifications.
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@ -1,12 +0,0 @@
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Subject: bridge_builtin_features
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Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
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Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
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interval in seconds will result in a periodic beep being
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played to the monitored channel upon MixMontior/Monitor
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feature start.
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If an interval less than 5 seconds is specified, the interval
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will default to 5 seconds. If the value is set to an invalid
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interval, the default of 15 seconds will be used.
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@ -1,6 +0,0 @@
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Subject: cdr
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Two new options have been added which allow
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bridging and dial state changes to be ignored
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in CDRs, which can be useful if a single CDR
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is desired for a channel.
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Subject: cli
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Subject: core
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This change increases the display width on 'core show channels'
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amd 'core show channels verbose'
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For 'core show channels', the Channel name field is increased to
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64 characters and the Location name field is increased to 32
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characters.
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For 'core show channels verbose', the Channel name field is
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increased to 80 characters, the Context is increased to 24
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characters and the Extension is increased to 24 characters.
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@ -1,5 +0,0 @@
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Subject: db
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The DBPrefixGet AMI action now allows retrieving
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all of the DB keys beginning with a particular
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prefix.
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Subject: DUNDi
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DUNDi now supports chan_pjsip. Outgoing calls using
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PJSIP require the pjsip_outgoing_endpoint option
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to be set in dundi.conf.
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Subject: features
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The Bridge application now has the n "no answer" option
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that can be used to prevent the channel from being
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automatically answered prior to bridging.
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Subject: format_sln
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format_sln now recognizes '.slin' as a valid
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file extension in addition to the existing
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'.sln' and '.raw'.
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Subject: New EXPORT function
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A new function, EXPORT, allows writing variables
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and functions on other channels, the complement
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of the IMPORT function.
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@ -1,5 +0,0 @@
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Subject: func_json
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Additional parsing capabilities have been added to the
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JSON_DECODE function, including support for arrays
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and recursive indexing.
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Subject: func_strings
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Three new functions, TRIM, LTRIM, and RTRIM, are
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now available for trimming leading and trailing
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whitespace.
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Subject: chan_dahdi
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FXO channels (FXS signaled) that don't use callerid or
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distinctive ring detection can now be configured
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to enter the dialplan immediately using immediate=yes,
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instead of waiting for at least one ring.
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Subject: http
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Master-Only: True
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For bound addresses, the HTTP status page now combines the bound
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address and bound port in a single line. Additionally, the SSL bind
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address has been renamed to TLS.
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@ -1,5 +0,0 @@
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Subject: locks
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A new AMI event, DeadlockStart, is now available
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when Asterisk is compiled with DETECT_DEADLOCKS,
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and can indicate that a deadlock has occured.
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Subject: AMI
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The AOCMessage action can now be used to generate AOC-S messages.
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Subject: res_geolocation
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* Added processing for the 'confidence' element.
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* Added documentation to some APIs.
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* removed a lot of complex code related to the very-off-nominal
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case of needing to process multiple location info sources.
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* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
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one eprofile instead of a datastore of multiples.
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* Plugged a huge leak in XML processing that arose from
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insufficient documentation by the libxml/libxslt authors.
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* Refactored stylesheets to be more efficient.
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* Renamed 'profile_action' to 'profile_precedence' to better
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reflect it's purpose.
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* Added the config option for 'allow_routing_use' which
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sets the value of the 'Geolocation-Routing' header.
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* Removed the GeolocProfileCreate and GeolocProfileDelete
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dialplan apps.
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* Changed the GEOLOC_PROFILE dialplan function as follows:
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* Removed the 'profile' argument.
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* Automatically create a profile if it doesn't exist.
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* Delete a profile if 'inheritable' is set to no.
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* Fixed various bugs and leaks
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* Updated Asterisk WiKi documentation.
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Added 4 built-in profiles:
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"<prefer_config>"
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"<discard_config>"
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"<prefer_incoming>"
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"<discard_incoming>"
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The profiles are empty except for having their precedence
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set.
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Added profile parameter "suppress_empty_ca_elements" that
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will cause Civic Address elements that are empty to be
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suppressed from the outgoing PIDF-LO document.
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You can now specify the location object's format, location_info,
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method, location_source and confidence parameters directly on
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a profile object for simple scenarios where the location
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information isn't common with any other profiles. This is
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mutually exclusive with setting location_reference on the
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profile.
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Added an 'a' option to the GEOLOC_PROFILE function to allow
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variable lists like location_info_refinement to be appended
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to instead of replacing the entire list.
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Added an 'r' option to the GEOLOC_PROFILE function to resolve all
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variables before a read operation and after a Set operation.
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@ -1,5 +0,0 @@
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Subject: Add support for named capture agent.
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A name for the capture agent can now be specified
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using the capture_name option which, if specified,
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will be sent to the HEP server.
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@ -1,12 +0,0 @@
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Subject: res_http_media_cache
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The res_http_media_cache module now attempts to load
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configuration from the res_http_media_cache.conf file.
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The following options were added:
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* timeout_secs
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* user_agent
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* follow_location
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* max_redirects
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* protocols
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* redirect_protocols
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* dns_cache_timeout_secs
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@ -1,4 +0,0 @@
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Subject: res_musiconhold_answeredonly
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This change adds an option, answeredonly, that will prevent music
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on hold on channels that are not answered.
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@ -1,5 +0,0 @@
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Subject: res_phoneprov
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On multihomed Asterisk servers with dynamic SERVER template variables,
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reloading this module is no longer required when re-provisioning your
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phone to another interface address (e.g. when moving between VLANs.)
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@ -1,6 +0,0 @@
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Subject: res_pjsip
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A new option named "peer_supported" has been added to the endpoint option
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100rel. When set to this option, Asterisk sends provisional responses
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reliably if the peer supports it. If the peer does not support reliable
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provisional responses, Asterisk sends them normally.
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@ -1,8 +0,0 @@
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Subject: res_pjsip
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A new option named "all_codecs_on_empty_reinvite" has been added to the
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global section. When this option is enabled, on reception of a re-INVITE
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without SDP, Asterisk will send an SDP offer in the 200 OK response containing
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all configured codecs on the endpoint, instead of simply those that have
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already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
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The default value is "off".
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@ -1,4 +0,0 @@
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Subject: res_pjsip_aoc
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Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
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A new endpoint option, send_aoc, controls this.
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@ -1,7 +0,0 @@
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Subject: res_pjsip_logger
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SIP messages can now be filtered by SIP request method
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(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
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SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
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allowing for more granular debugging to be done
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in the CLI. This applies to requests but not responses.
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@ -1,4 +0,0 @@
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Subject: res_pjsip_notify
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Allows using the config options in pjsip_notify.conf
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from AMI actions as with the existing CLI commands.
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@ -1,5 +0,0 @@
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Subject: res_pjsip_header_funcs
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The new PJSIP_HEADER_PARAM function now fully supports both
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URI and header parameters. Both reading and writing
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parameters are supported.
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@ -1,6 +0,0 @@
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Subject: res_pjsip
|
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|
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Added options "security_negotiation" and "security_mechanisms" to pjsip
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endpoints and registrations. "security_negotiation" can be set to "no" (default)
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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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@ -1,4 +0,0 @@
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Subject: res_pjsip_session
|
||||
|
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The overlap_context option now allows explicitly
|
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specifying a context to use for overlap dialing matches.
|
||||
|
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@ -1,5 +0,0 @@
|
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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
|
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changed.
|
||||
|
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@ -1,4 +0,0 @@
|
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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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@ -1,9 +0,0 @@
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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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||||
|
|
@ -1,5 +0,0 @@
|
|||
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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: res_tonedetect
|
||||
|
||||
The TONE_DETECT function now supports
|
||||
detection of audible ringback tone
|
||||
using the p option.
|
||||
|
|
@ -1,11 +0,0 @@
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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.
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||||
|
||||
The "failures" attribute was added to the "testsuite"
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||||
element.
|
||||
|
||||
Also added two new unit tests that just pass and fail
|
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to be used for testing CI itself.
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|
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@ -1,14 +0,0 @@
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Subject: Transfer feature
|
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|
||||
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).
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
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.
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
## **DO NOT REMOVE THIS FILE!**
|
||||
|
||||
The only files that should be added to this directory are ones that will be
|
||||
used by the release script to update the UPGRADE.txt file automatically. The
|
||||
only time that it is necessary to add something to the UPGRADE-staging directory
|
||||
is if you are making a breaking change to an existing feature in Asterisk. The
|
||||
file does not need to have a meaningful name, but it probably should. If there
|
||||
are multiple items that need documenting, you can add multiple files, each with
|
||||
their own description. If the message is going to be the same for each subject,
|
||||
then you can add multiple subject headers to one file. The "Subject: xxx" line
|
||||
is case sensitive! For example, if you are making a change to PJSIP, then you
|
||||
might add the file "res_pjsip_my_cool_feature.txt" to this directory, with a
|
||||
short description of what it does. The files must have the ".txt" suffix.
|
||||
If you are adding multiple entries, they should be done in the same commit
|
||||
to avoid merge conflicts. Here's an example:
|
||||
|
||||
> Subject: res_pjsip
|
||||
> Subject: Core
|
||||
>
|
||||
> Here's a pretty good description of my new feature that explains exactly what
|
||||
> it does and how to use it.
|
||||
|
||||
Here's a master-only example:
|
||||
|
||||
> Subject: res_ari
|
||||
> Master-Only: True
|
||||
>
|
||||
> This change will only go into the master branch. The "Master-Only" header
|
||||
> will never be in a change not in master.
|
||||
|
||||
Note that the second subject has another header: "Master-Only". Changes that go
|
||||
into the master branch and ONLY the master branch are the only ones that should
|
||||
have this header. Also, the value can only be "true" or "True". The
|
||||
"Master-Only" part of the header IS case-sensitive, however!
|
||||
|
||||
For more information, check out the wiki page:
|
||||
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: app_cdr
|
||||
Master-Only: True
|
||||
|
||||
The previously deprecated NoCDR application has been removed.
|
||||
Additionally, the previously deprecated 'e' option to the ResetCDR
|
||||
application has been removed.
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
Subject: app_macro
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 16
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
||||
|
||||
For most modules that interacted with app_macro,
|
||||
this change is limited to no longer looking for
|
||||
the current context from the macrocontext when set.
|
||||
|
||||
The following modules have additional impacts:
|
||||
|
||||
app_dial - no longer supports M^ connected/redirecting macro
|
||||
|
||||
app_minivm - samples written using macro will no longer work.
|
||||
The sample needs to be re-written
|
||||
|
||||
app_queue - can no longer call a macro on the called party's
|
||||
channel. Use gosub which is currently supported
|
||||
|
||||
ccss - no callback macro, gosub only
|
||||
|
||||
app_voicemail - no macro support
|
||||
|
||||
channel - remove macrocontext and priority, no connected
|
||||
line or redirection macro options
|
||||
|
||||
options - stdexten is deprecated to gosub as the default
|
||||
and only options
|
||||
|
||||
pbx - removed macrolock
|
||||
|
||||
pbx_dundi - no longer look for macro
|
||||
|
||||
snmp - removed macro context, exten, and priority
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: app_osplookup
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 19
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: chan_alsa
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 19
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: chan_mgcp
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 19
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: chan_sip
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 17
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: chan_skinny
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 19
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: pbx_builtins
|
||||
Master-Only: True
|
||||
|
||||
The previously deprecated ImportVar and SetAMAFlags
|
||||
applications have now been removed.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
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.
|
||||
|
|
@ -1,13 +0,0 @@
|
|||
Subject: res_monitor
|
||||
Master-Only: True
|
||||
|
||||
This module was deprecated in Asterisk 16
|
||||
and is now being removed in accordance with
|
||||
the Asterisk Module Deprecation policy.
|
||||
|
||||
This also removes the 'w' and 'W' options
|
||||
for app_queue.
|
||||
|
||||
MixMonitor should be default and only option
|
||||
for all settings that previously used either
|
||||
Monitor or MixMonitor.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: translate.c
|
||||
Master-Only: True
|
||||
|
||||
When setting up translation between two codecs the quality was not taken into account,
|
||||
resulting in suboptimal translation. The quality is now taken into account,
|
||||
which can reduce the number of translation steps required, and improve the resulting quality.
|
||||
|
|
@ -54,7 +54,7 @@
|
|||
- \ref manager.c Main manager code file
|
||||
*/
|
||||
|
||||
#define AMI_VERSION "10.0.0"
|
||||
#define AMI_VERSION "11.0.0"
|
||||
#define DEFAULT_MANAGER_PORT 5038 /* Default port for Asterisk management via TCP */
|
||||
#define DEFAULT_MANAGER_TLS_PORT 5039 /* Default port for Asterisk management via TCP */
|
||||
|
||||
|
|
|
|||
|
|
@ -58,7 +58,6 @@ struct ast_ari_endpoints_send_message_args {
|
|||
const char *from;
|
||||
/*! The body of the message */
|
||||
const char *body;
|
||||
/*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */
|
||||
struct ast_json *variables;
|
||||
};
|
||||
/*!
|
||||
|
|
@ -150,7 +149,6 @@ struct ast_ari_endpoints_send_message_to_endpoint_args {
|
|||
const char *from;
|
||||
/*! The body of the message */
|
||||
const char *body;
|
||||
/*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */
|
||||
struct ast_json *variables;
|
||||
};
|
||||
/*!
|
||||
|
|
|
|||
|
|
@ -2,7 +2,7 @@
|
|||
"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
|
||||
"_author": "David M. Lee, II <dlee@digium.com>",
|
||||
"_svn_revision": "$Revision$",
|
||||
"apiVersion": "9.0.0",
|
||||
"apiVersion": "10.0.0",
|
||||
"swaggerVersion": "1.1",
|
||||
"basePath": "http://localhost:8088/ari",
|
||||
"apis": [
|
||||
|
|
|
|||
Loading…
Reference in New Issue