mirror of https://github.com/asterisk/asterisk
George Joseph
3 years ago
committed by
Asterisk Development Team
parent
68349125a8
commit
f7ea380b04
@ -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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@ -1,5 +0,0 @@
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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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@ -1,6 +0,0 @@
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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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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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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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@ -1,14 +0,0 @@
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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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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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@ -1,5 +0,0 @@
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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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@ -1,5 +0,0 @@
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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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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: 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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@ -1,49 +0,0 @@
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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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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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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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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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Subject: res_pjsip
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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
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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
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TLS transports in res_pjsip can now reload their TLS certificate
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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
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user_eq_phone=yes flag on a pjsip endpoint will now set user=phone on
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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
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This module has been updated to provide additional
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quality statistics in the form of an Asterisk
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Media Experience Score. The score is available using
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the same mechanisms you'd use to retrieve jitter, loss,
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and rtt statistics. For more information about the
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score and how to retrieve it, see
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https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
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@ -1,5 +0,0 @@
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Subject: res_pjsip_rfc3326
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Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
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addition to currently supported Q.850). The first header found will be used to set
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the HANGUPCAUSE variable.
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@ -1,5 +0,0 @@
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Subject: res_tonedetect
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The TONE_DETECT function now supports
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detection of audible ringback tone
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using the p option.
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@ -1,11 +0,0 @@
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Subject: test.c
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The "tests" attribute of the "testsuite" element in the
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output XML now reflects only the tests actually requested
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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.
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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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@ -1,14 +0,0 @@
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Subject: Transfer feature
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The following capabilities have been added to the
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transfer feature:
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- The transfer initiation announcement prompt can
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now be customized in features.conf.
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- The TRANSFER_EXTEN variable now can be set on the
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transferer's channel in order to allow the transfer
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function to automatically attempt to go to the extension
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contained in this variable, if it exists. The transfer
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context behavior is not changed (TRANSFER_CONTEXT is used
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if it exists; otherwise the default context is used).
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@ -1,5 +0,0 @@
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Subject: xmldocs
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The XML documentation can now be reloaded without restarting
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Asterisk, which makes it possible to load new modules that
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enforce documentation without restarting Asterisk.
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@ -1,8 +0,0 @@
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Subject: app_playback
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In Asterisk 11, if a channel was redirected away during Playback(),
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the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
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(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
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behavior was inadvertently changed and the same operation would result
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in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
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behavior has been restored.
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@ -1,8 +0,0 @@
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Subject: AMI (Asterisk Manager Interface)
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Previously, GetConfig and UpdateConfig were able to access files outside of
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the Asterisk configuration directory. Now this access is put behind the
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live_dangerously configuration option in asterisk.conf, which is disabled by
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default. If access to configuration files outside of the Asterisk configuation
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directory is required via AMI, then the live_dangerously configuration option
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must be set to yes.
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@ -1,5 +0,0 @@
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Subject: res_crypto
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In addition to only paying attention to files ending with .key or .pub
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in the keys directory, we now also ignore any files which aren't regular
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files.
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Loading…
Reference in new issue