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@ -12,6 +12,384 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- New functionality introduced in Asterisk 18.0.0 --------------------------
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------------------------------------------------------------------------------
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Core
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------------------
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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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app_bridgeaddchan
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------------------
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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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res_pjsip
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------------------
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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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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
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------------------------------------------------------------------------------
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AMI
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------------------
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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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ARI
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------------------
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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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ARI Channels
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------------------
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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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Core
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------------------
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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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Features
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------------------
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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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Messaging
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------------------
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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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STIR/SHAKEN
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------------------
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* STIR/SHAKEN support has been added to Asterisk. Configuration is done in
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stir_shaken.conf. There is a sample configuration file to help you get
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started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
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set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
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to yes on the endpoint configuration object. This will add an Identity
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header on outgoing INVITEs, and check for an Identity header on incoming
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INVITEs. This option has been added to Alembic as well.
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The information received on an incoming INVITE can be checked using the
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STIR_SHAKEN dialplan function. There are two variations:
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STIR_SHAKEN(count)
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STIR_SHAKEN(0, verify_result)
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The first variation will tell you how many STIR/SHAKEN results are on the
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channel. The second fetches information for a specific result. The first
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parameter is the index, followed by what information you want to retrieve.
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The available options are 'verify_result', 'identity', and 'attestation'.
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app_chanisavail
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------------------
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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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app_confbridge
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------------------
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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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* 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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app_dial
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------------------
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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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app_mixmonitor
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------------------
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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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* 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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app_page
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------------------
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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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app_voicemail
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------------------
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* A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
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the Asterisk voicemail directory on startup. Some users that store their
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voicemails on network storage devices experienced slow startup times due to the
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relative expense of traversing the voicemail directory structure looking for
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orphaned lock files. This feature has now been removed.
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Users who require the lock files to be removed at startup should modify their
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startup scripts to do so before starting the asterisk process.
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chan_pjsip
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------------------
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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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chan_rtp
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------------------
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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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func_curl
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------------------
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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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* 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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func_jitterbuffer
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------------------
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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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func_volume
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------------------
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* Accept decimal number as argument.
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http
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------------------
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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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minmemfree
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------------------
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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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res_ari_channels
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------------------
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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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res_musiconhold
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------------------
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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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* 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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res_pjsip
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------------------
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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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res_pjsip_endpoint_identifier_ip
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------------------
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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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res_pjsip_logger
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------------------
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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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res_pjsip_session
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------------------
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* When placing an outgoing call to a PJSIP endpoint the intent
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of any requested formats will now be respected. If only an audio
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format is requested (such as ulaw) but the underlying endpoint
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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
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video.
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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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res_rtp_asterisk
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------------------
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* This change include a new cli command 'rtp show settings'
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The command display by general settings of rtp configuration. For this
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point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
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strictrtp, learning_min_sequential and icesupport.
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* The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
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an ACL mechanism.
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As such six now options are now available:
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ice_deny
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ice_permit
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ice_acl
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stun_deny
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stun_permit
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stun_acl
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These options have their obvious meanings as used elsewhere.
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Backwards compatibility was maintained by adding {stun,ice}_blacklist as
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aliases for {stun,ice}_deny.
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res_sorcery_memory_cache
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------------------
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* The SorceryMemoryCacheExpireObject AMI action and CLI
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command allow expiring of a specific object within the
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sorcery memory cache. This is done by removing the
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object from the cache with the expectation that the
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cache will then re-populate the object when it is next
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needed.
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For full backend caching this does not occur. The cache
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won't repopulate until an entire refresh is done resulting
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in the possibility that objects are missing until that
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time.
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The AMI action and CLI command will now not allow
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expiring of an object if the cache is configured as a
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full backend cache. Instead you must use either the
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SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
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AMI actions or their associated CLI commands.
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|
|
taskprocessor.c
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|
|
------------------
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* Added two new CLI commands to reset stats for taskprocessors. You can
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reset stats for a single, specific taskprocessor ('core reset
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|
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
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('core reset taskprocessors'). These commands will reset the counter for
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|
the number of tasks processed as well as the max queue size.
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* Added "like" support for 'core show taskprocessors'. Now you
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can specify a specific set of taskprocessors (or just one) by
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|
|
adding the keyword "like" to the above command, followed by
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|
|
your search criteria.
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|
------------------------------------------------------------------------------
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--- New functionality introduced in Asterisk 17.0.0 --------------------------
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------------------------------------------------------------------------------
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Asterisk Development Team
5 years ago