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Update CHANGES and UPGRADE.txt for 18.0.0
This commit is contained in:
Asterisk Development Team
@@ -1,44 +0,0 @@
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Subject: Core
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Master-Only: True
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The Streams API becomes the home for the core ACN capabilities.
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These include...
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* Parsing and formatting of codec negotation preferences.
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* Resolving pending streams and topologies with those configured
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using configured preferences.
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* Utility functions for creating string representations of
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streams, topologies, and negotiation preferences.
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For codec negotiation preferences:
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* Added ast_stream_codec_prefs_parse() which takes a string
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representation of codec negotiation preferences, which
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may come from a pjsip endpoint for example, and populates
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a ast_stream_codec_negotiation_prefs structure.
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* Added ast_stream_codec_prefs_to_str() which does the reverse.
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* Added many functions to parse individual parameter name
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and value strings to their respectrive enum values, and the
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reverse.
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For streams:
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* Added ast_stream_create_resolved() which takes a "live" stream
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and resolves it with a configured stream and the negotiation
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preferences to create a new stream.
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* Added ast_stream_to_str() which create a string representation
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of a stream suitable for debug or display purposes.
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For topology:
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* Added ast_stream_topology_create_resolved() which takes a "live"
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topology and resolves it, stream by stream, with a configured
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topology stream and the negotiation preferences to create a new
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topology.
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* Added ast_stream_topology_to_str() which create a string
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representation of a topology suitable for debug or display
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purposes.
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* Renamed ast_format_caps_from_topology() to
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ast_stream_topology_get_formats() to be more consistent with
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the existing ast_stream_get_formats().
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Additional changes:
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* A new function ast_format_cap_append_names() appends the results
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to the ast_str buffer instead of replacing buffer contents.
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@@ -1,10 +0,0 @@
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Subject: ARI Channels
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The Channel resource has a new sub-resource "externalMedia".
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This allows an application to create a channel for the sole purpose
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of exchanging media with an external server. Once created, this
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channel could be placed into a bridge with existing channels to
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allow the external server to inject audio into the bridge or
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receive audio from the bridge.
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See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
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for more information.
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@@ -1,4 +0,0 @@
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Subject: AMI
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You can now specify an optional 'Content-Type' as an argument for the Asterisk
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SendText manager action.
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@@ -1,7 +0,0 @@
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Subject: app_bridgeaddchan
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Master-Only: true
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The BridgeAdd application now behaves more like the Bridge application.
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The application now sets the BRIDGERESULT channel variable to indicate
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what happened when the channel resumes in dialplan. This is instead of
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hanging up the channel on failure conditions.
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@@ -1,5 +0,0 @@
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Subject: app_chanisavail
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The ChanIsAvail application now tolerates empty positions in the supplied
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device list. Dialplan can now be simplified by not having to check for
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empty positions in the device list.
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@@ -1,5 +0,0 @@
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Subject: app_confbridge
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A new bridge profile option, maximum_sample_rate, has been added which sets
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a maximum sample rate that the bridge will be mixed at. This allows the bridge
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to move below the maximum sample rate as needed but caps it at the maximum.
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@@ -1,6 +0,0 @@
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Subject: app_dial
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The Dial application now tolerates empty positions in the supplied
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destination list. Dialplan can now be simplified by not having to check
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for empty positions in the destination list. If there are no endpoints to
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dial then DIALSTATUS is set to CHANUNAVAIL.
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@@ -1,8 +0,0 @@
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Subject: app_mixmonitor
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If the 'filename' argument to MixMonitor() ended with '.wav49,'
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Asterisk would silently convert the extension to '.WAV' when opening
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the file for writing. This caused the MIXMONITOR_FILENAME variable to
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reference the wrong file. The MIXMONITOR_FILENAME variable will now
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reflect the name of the file that Asterisk actually used instead of
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the filename that was passed to the application.
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@@ -1,5 +0,0 @@
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Subject: app_page
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The Page application now tolerates empty positions in the supplied
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destination list. Dialplan can now be simplified by not having to check
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for empty positions in the destination list.
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@@ -1,5 +0,0 @@
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Subject: ARI
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A new parameter 'inhibitConnectedLineUpdates' is now available in the
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'bridges.addChannel' call. This prevents the identity of the newly connected
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channel from being presented to other bridge members.
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@@ -1,5 +0,0 @@
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Subject: res_ari_channels
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When creating a channel in ARI using the create call
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you can now specify dialplan variables to be set as part
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of the same operation.
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@@ -1,5 +0,0 @@
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Subject: chan_pjsip
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A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This
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allows the behaviour of the moh_passthrough endpoint option to be read or changed
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in the dialplan. This allows control on a per-call basis.
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@@ -1,7 +0,0 @@
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Subject: chan_rtp
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The UnicastRTP channel driver provided by chan_rtp now accepts
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"<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
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The first AAAA (preferred) or A record resolved will be used as the destination.
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The lookup is synchronous so beware of possible dialplan delays if you specify a
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hostname.
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@@ -1,7 +0,0 @@
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Subject: app_confbridge
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A new option, "text_messaging", has been added to the user profile
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which allows control over whether text messaging is enabled or
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disabled for a user. If enabled (the default) text messages
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will be sent to the user. If disabled no text messages will be
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sent to the user.
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@@ -1,14 +0,0 @@
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Subject: Features
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Adds support for AudioSocket, a very simple bidirectional audio streaming
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protocol. There are both channel and application interfaces.
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A description of the protocol can be found on the referenced wiki page. A
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short talk about the reasons and implementation can be found on YouTube at
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the link provided.
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ARI support has also been added via the existing "externalMedia" ARI
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functionality. The UUID is specified using the arbitrary "data" field.
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Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
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YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
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@@ -1,5 +0,0 @@
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Subject: func_curl
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A new option, followlocation, can now be enabled with the CURLOPT()
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dialplan function. Setting this will instruct cURL to follow 3xx
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redirects, which it does not by default.
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@@ -1,6 +0,0 @@
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Subject: func_curl
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A new parameter, httpheader, has been added to CURLOPT function. This parameter
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allows to set custom http headers for subsequent calls off CURL function.
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Any setting of headers will replace the default curl headers
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(e.g. "Content-type: application/x-www-form-urlencoded")
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@@ -1,6 +0,0 @@
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Subject: func_jitterbuffer
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The JITTERBUFFER dialplan function now has an option to enable video synchronization
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support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
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the video is buffered according to the size of the audio jitterbuffer and is
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synchronized to the audio.
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@@ -1,3 +0,0 @@
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Subject: func_volume
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Accept decimal number as argument.
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@@ -1,6 +0,0 @@
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Subject: Core
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H.265/HEVC is now a supported video codec and it can be used by
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specifying "h265" in the allow line.
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Please note however, that handling of the additional SDP parameters
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described in RFC 7798 section 7.2 is not yet supported.
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@@ -1,11 +0,0 @@
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Subject: Messaging
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In order to reduce the amount of AMI and ARI events generated,
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the global "Message/ast_msg_queue" channel can be set to suppress
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it's normal channel housekeeping events such as "Newexten",
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"VarSet", etc. This can greatly reduce load on the manager
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and ARI applications when the Digium Phone Module for Asterisk
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is in use. To enable, set "hide_messaging_ami_events" in
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asterisk.conf to "yes" In Asterisk versions <18, the default
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is "no" preserving existing behavior. Beginning with
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Asterisk 18, the option will default to "yes".
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@@ -1,4 +0,0 @@
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Subject: http
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You can now disable the /httpstatus page served by Asterisk's built-in
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HTTP server by setting 'enable_status' to 'no' in http.conf.
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@@ -1,5 +0,0 @@
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Subject: minmemfree
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The 'minmemfree' configuration option now counts memory allocated to
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the filesystem cache as "free" because it is memory that is available
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to the process.
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@@ -1,7 +0,0 @@
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Subject: app_mixmonitor
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An option 'S' has been added to MixMonitor. If used in combination with
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the r() and/or t() options, if a frame is available to write to one of
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those files but not the other, a frame of silence if written to the file
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that does not have an audio frame. This should prevent the two files
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from "drifting" when mixed after the fact.
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@@ -1,5 +0,0 @@
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Subject: res_musiconhold
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A new mode - playlist - has been added to res_musiconhold. This mode allows the
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user to specify the files (or URLs) to play explicitly by putting them directly
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in musiconhold.conf.
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@@ -1,21 +0,0 @@
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Subject: res_pjsip_logger
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The PJSIP packet logger now has the following CLI commands:
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pjsip set logger pcap <filename>
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When used this will create a pcap file containing the incoming
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and outgoing SIP packets, in unencrypted form.
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pjsip set logger console <on / off>
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This allows you to toggle logging to console on and off.
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pjsip set logger host <IP/subnet mask> add
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This allows you to add an additional IP address or subnet
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mask to logging, allowing you to log multiple instead of
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just a single IP address or all traffic.
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The normal "pjsip set logger host" CLI command has also been
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expanded to allow subnet masks as well.
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@@ -1,7 +0,0 @@
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Subject: res_musiconhold
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This fix allows a realtime moh class to be unregistered from the command
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line. This is useful when the contents of a directory referenced by a
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realtime moh class have changed.
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The realtime moh class is then reloaded on the next request and uses the
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new directory contents.
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@@ -1,9 +0,0 @@
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Subject: res_pjsip
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Added a new PJSIP system setting called disable_rport.
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Default is no to keep support working as before.
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If it is false (default) it adds the 'rport' parameter in the outgoing request message.
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If it is true it does not add the 'rport' parameter in the outgoing request message.
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This is a system option, but working as a global option.
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@@ -1,8 +0,0 @@
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Subject: res_pjsip
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Subject: res_pjsip_session
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Master-Only: True
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Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
|
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have been added to res_pjsip endpoints that specify the preferred order
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of codecs to use between those received/sent in an SDP offer and those
|
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set in the endpoint configuration.
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@@ -1,8 +0,0 @@
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Subject: res_pjsip_endpoint_identifier_ip
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In 'type = identify' sections, the addresses specified for the 'match'
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clause can now include a port number. For IP addresses, the port is
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provided by including a colon after the address, followed by the
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desired port number. If supplied, the netmask should follow the port
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number. To specify a port for IPv6 addresses, the address itself must
|
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be enclosed in brackets to be parsed correctly.
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@@ -1,8 +0,0 @@
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Subject: res_pjsip_session
|
||||
|
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When placing an outgoing call to a PJSIP endpoint the intent
|
||||
of any requested formats will now be respected. If only an audio
|
||||
format is requested (such as ulaw) but the underlying endpoint
|
||||
does not support the format the resulting SDP will still only
|
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contain an audio stream, and not any additional streams such as
|
||||
video.
|
||||
@@ -1,18 +0,0 @@
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Subject: res_rtp_asterisk
|
||||
|
||||
The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
|
||||
an ACL mechanism.
|
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|
||||
As such six now options are now available:
|
||||
|
||||
ice_deny
|
||||
ice_permit
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||||
ice_acl
|
||||
stun_deny
|
||||
stun_permit
|
||||
stun_acl
|
||||
|
||||
These options have their obvious meanings as used elsewhere.
|
||||
|
||||
Backwards compatibility was maintained by adding {stun,ice}_blacklist as
|
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aliases for {stun,ice}_deny.
|
||||
@@ -1,7 +0,0 @@
|
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Subject: res_rtp_asterisk
|
||||
|
||||
This change include a new cli command 'rtp show settings'
|
||||
|
||||
The command display by general settings of rtp configuration. For this
|
||||
point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
|
||||
strictrtp, learning_min_sequential and icesupport.
|
||||
@@ -1,19 +0,0 @@
|
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Subject: res_sorcery_memory_cache
|
||||
|
||||
The SorceryMemoryCacheExpireObject AMI action and CLI
|
||||
command allow expiring of a specific object within the
|
||||
sorcery memory cache. This is done by removing the
|
||||
object from the cache with the expectation that the
|
||||
cache will then re-populate the object when it is next
|
||||
needed.
|
||||
|
||||
For full backend caching this does not occur. The cache
|
||||
won't repopulate until an entire refresh is done resulting
|
||||
in the possibility that objects are missing until that
|
||||
time.
|
||||
|
||||
The AMI action and CLI command will now not allow
|
||||
expiring of an object if the cache is configured as a
|
||||
full backend cache. Instead you must use either the
|
||||
SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
|
||||
AMI actions or their associated CLI commands.
|
||||
@@ -1,20 +0,0 @@
|
||||
Subject: STIR/SHAKEN
|
||||
|
||||
STIR/SHAKEN support has been added to Asterisk. Configuration is done in
|
||||
stir_shaken.conf. There is a sample configuration file to help you get
|
||||
started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
|
||||
set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
|
||||
to yes on the endpoint configuration object. This will add an Identity
|
||||
header on outgoing INVITEs, and check for an Identity header on incoming
|
||||
INVITEs. This option has been added to Alembic as well.
|
||||
|
||||
The information received on an incoming INVITE can be checked using the
|
||||
STIR_SHAKEN dialplan function. There are two variations:
|
||||
|
||||
STIR_SHAKEN(count)
|
||||
STIR_SHAKEN(0, verify_result)
|
||||
|
||||
The first variation will tell you how many STIR/SHAKEN results are on the
|
||||
channel. The second fetches information for a specific result. The first
|
||||
parameter is the index, followed by what information you want to retrieve.
|
||||
The available options are 'verify_result', 'identity', and 'attestation'.
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: taskprocessor.c
|
||||
|
||||
Added "like" support for 'core show taskprocessors'. Now you
|
||||
can specify a specific set of taskprocessors (or just one) by
|
||||
adding the keyword "like" to the above command, followed by
|
||||
your search criteria.
|
||||
@@ -1,7 +0,0 @@
|
||||
Subject: taskprocessor.c
|
||||
|
||||
Added two new CLI commands to reset stats for taskprocessors. You can
|
||||
reset stats for a single, specific taskprocessor ('core reset
|
||||
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
|
||||
('core reset taskprocessors'). These commands will reset the counter for
|
||||
the number of tasks processed as well as the max queue size.
|
||||
@@ -1,10 +0,0 @@
|
||||
Subject: app_voicemail
|
||||
|
||||
A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
|
||||
the Asterisk voicemail directory on startup. Some users that store their
|
||||
voicemails on network storage devices experienced slow startup times due to the
|
||||
relative expense of traversing the voicemail directory structure looking for
|
||||
orphaned lock files. This feature has now been removed.
|
||||
|
||||
Users who require the lock files to be removed at startup should modify their
|
||||
startup scripts to do so before starting the asterisk process.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: Core
|
||||
Master-Only: True
|
||||
|
||||
The ast_format_cap_from_stream_topology() function has been renamed
|
||||
to ast_stream_topology_get_formats().
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: AMI
|
||||
|
||||
The AMI Originate action, which optionally takes a dialplan application as
|
||||
an argument, no longer accepts "Originate" as the application due to
|
||||
security concerns.
|
||||
@@ -1,7 +0,0 @@
|
||||
Subject: app_bridgeaddchan
|
||||
Master-Only: true
|
||||
|
||||
The BridgeAdd application now behaves more like the Bridge application.
|
||||
The application now sets the BRIDGERESULT channel variable to indicate
|
||||
what happened when the channel resumes in dialplan. This is instead of
|
||||
hanging up the channel on failure conditions.
|
||||
@@ -1,10 +0,0 @@
|
||||
Subject: app_mixmonitor
|
||||
Master-Only: true
|
||||
|
||||
In Asterisk 13.29, a new option flag was added to MixMonitor (the 'S'
|
||||
option) that when combined with the r() or t() options would inject
|
||||
silence into these files if audio was going to be written to one and
|
||||
not that other. This allowed the files specified by r() and t() to
|
||||
subsequently be mixed outside of Asterisk and be appropriately
|
||||
synchronized. This behavior is now the default, and a new option has
|
||||
been added to disable this behavior if desired (the 'n' option).
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: app_queue
|
||||
Master-Only: true
|
||||
|
||||
If they are not specified in [general], "shared_lastcall" and "autofill"
|
||||
now always default to OFF. Before this version, they would be off ('no') if
|
||||
queues.conf did not have a [general] section, but on ('yes') if it did.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: app_queue
|
||||
Master-Only: True
|
||||
|
||||
The 'Reason' header in the QueueMemberPause AMI Event has been
|
||||
removed. The 'PausedReason' header should be used instead.
|
||||
@@ -1,26 +0,0 @@
|
||||
Subject: ARI
|
||||
Subject: res_stasis
|
||||
|
||||
The "TextMessageReceived" event used to include a list of "TextMessageVariable"
|
||||
objects as part of its output. Due to a couple of bugs in Asterisk a list of
|
||||
received variables was never included even if ones were available. However,
|
||||
variables set to send would be (which they should have not been), but would
|
||||
fail validation due to the bad formatting.
|
||||
|
||||
So basically there was no way to get a "TextMessageReceived" event with
|
||||
variables. Due to this the API has changed. The "TextMessageVariable" object
|
||||
no longer exists. "TextMessageReceived" now returns a JSON object of key/value
|
||||
pairs. So for instance instead of a list of "TextMessageVariable" objects:
|
||||
|
||||
[ TextMessageVariable, TextMessageVariable, TextMessageVariable]
|
||||
|
||||
where a TextMessageVariable was supposed to be:
|
||||
|
||||
{ "key": "<var name>", "value":, "<var value>" }
|
||||
|
||||
The output is now just:
|
||||
|
||||
{ "<var name>": "<var value>" }
|
||||
|
||||
This aligns more with how variables are specified when sending a message, as
|
||||
well as other variable lists in ARI.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: res_stir_shaken
|
||||
|
||||
A new directory has been added under the default (e.g., /var/lib/asterisk) -
|
||||
inside the 'keys' directory - named 'stir_shaken'. This directory will
|
||||
hold public keys that have been downloaded for STIR/SHAKEN verification.
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: Core
|
||||
|
||||
The streams API function ast_stream_get_formats is
|
||||
now defined as returning the format capabilities const.
|
||||
This has always been the case but was never enforced
|
||||
through the API itself. Any consumer of this API that
|
||||
is not treating the formats as immutable should update
|
||||
their code to create a new format capabilities and set
|
||||
it on the stream instead.
|
||||
@@ -1,7 +0,0 @@
|
||||
Subject: app_voicemail
|
||||
Master-Only: True
|
||||
|
||||
The MessageExists dialplan application and the MESSAGE_EXISTS dialplan
|
||||
function were removed. The were deprecated in Asterisk 1.6.0 and
|
||||
Asterisk 11.0.0 respectively. The VM_INFO() dialplan function is the
|
||||
supported mechanism to query the status of a given mailbox.
|
||||
Reference in New Issue
Block a user