===
==============================================================================
+------------------------------------------------------------------------------
+--- New functionality introduced in Asterisk 18.0.0 --------------------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * The Streams API becomes the home for the core ACN capabilities.
+ These include...
+
+ * Parsing and formatting of codec negotation preferences.
+ * Resolving pending streams and topologies with those configured
+ using configured preferences.
+ * Utility functions for creating string representations of
+ streams, topologies, and negotiation preferences.
+
+ For codec negotiation preferences:
+ * Added ast_stream_codec_prefs_parse() which takes a string
+ representation of codec negotiation preferences, which
+ may come from a pjsip endpoint for example, and populates
+ a ast_stream_codec_negotiation_prefs structure.
+ * Added ast_stream_codec_prefs_to_str() which does the reverse.
+ * Added many functions to parse individual parameter name
+ and value strings to their respectrive enum values, and the
+ reverse.
+
+ For streams:
+ * Added ast_stream_create_resolved() which takes a "live" stream
+ and resolves it with a configured stream and the negotiation
+ preferences to create a new stream.
+ * Added ast_stream_to_str() which create a string representation
+ of a stream suitable for debug or display purposes.
+
+ For topology:
+ * Added ast_stream_topology_create_resolved() which takes a "live"
+ topology and resolves it, stream by stream, with a configured
+ topology stream and the negotiation preferences to create a new
+ topology.
+ * Added ast_stream_topology_to_str() which create a string
+ representation of a topology suitable for debug or display
+ purposes.
+ * Renamed ast_format_caps_from_topology() to
+ ast_stream_topology_get_formats() to be more consistent with
+ the existing ast_stream_get_formats().
+
+ Additional changes:
+ * A new function ast_format_cap_append_names() appends the results
+ to the ast_str buffer instead of replacing buffer contents.
+
+app_bridgeaddchan
+------------------
+ * 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.
+
+res_pjsip
+------------------
+ * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
+ have been added to res_pjsip endpoints that specify the preferred order
+ of codecs to use between those received/sent in an SDP offer and those
+ set in the endpoint configuration.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * You can now specify an optional 'Content-Type' as an argument for the Asterisk
+ SendText manager action.
+
+ARI
+------------------
+ * A new parameter 'inhibitConnectedLineUpdates' is now available in the
+ 'bridges.addChannel' call. This prevents the identity of the newly connected
+ channel from being presented to other bridge members.
+
+ARI Channels
+------------------
+ * The Channel resource has a new sub-resource "externalMedia".
+ This allows an application to create a channel for the sole purpose
+ of exchanging media with an external server. Once created, this
+ channel could be placed into a bridge with existing channels to
+ allow the external server to inject audio into the bridge or
+ receive audio from the bridge.
+ See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
+ for more information.
+
+Core
+------------------
+ * H.265/HEVC is now a supported video codec and it can be used by
+ specifying "h265" in the allow line.
+ Please note however, that handling of the additional SDP parameters
+ described in RFC 7798 section 7.2 is not yet supported.
+
+Features
+------------------
+ * Adds support for AudioSocket, a very simple bidirectional audio streaming
+ protocol. There are both channel and application interfaces.
+
+ A description of the protocol can be found on the referenced wiki page. A
+ short talk about the reasons and implementation can be found on YouTube at
+ the link provided.
+
+ ARI support has also been added via the existing "externalMedia" ARI
+ functionality. The UUID is specified using the arbitrary "data" field.
+
+ Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
+ YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
+
+Messaging
+------------------
+ * In order to reduce the amount of AMI and ARI events generated,
+ the global "Message/ast_msg_queue" channel can be set to suppress
+ it's normal channel housekeeping events such as "Newexten",
+ "VarSet", etc. This can greatly reduce load on the manager
+ and ARI applications when the Digium Phone Module for Asterisk
+ is in use. To enable, set "hide_messaging_ami_events" in
+ asterisk.conf to "yes" In Asterisk versions <18, the default
+ is "no" preserving existing behavior. Beginning with
+ Asterisk 18, the option will default to "yes".
+
+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'.
+
+app_chanisavail
+------------------
+ * The ChanIsAvail application now tolerates empty positions in the supplied
+ device list. Dialplan can now be simplified by not having to check for
+ empty positions in the device list.
+
+app_confbridge
+------------------
+ * A new bridge profile option, maximum_sample_rate, has been added which sets
+ a maximum sample rate that the bridge will be mixed at. This allows the bridge
+ to move below the maximum sample rate as needed but caps it at the maximum.
+
+ * A new option, "text_messaging", has been added to the user profile
+ which allows control over whether text messaging is enabled or
+ disabled for a user. If enabled (the default) text messages
+ will be sent to the user. If disabled no text messages will be
+ sent to the user.
+
+app_dial
+------------------
+ * The Dial application now tolerates empty positions in the supplied
+ destination list. Dialplan can now be simplified by not having to check
+ for empty positions in the destination list. If there are no endpoints to
+ dial then DIALSTATUS is set to CHANUNAVAIL.
+
+app_mixmonitor
+------------------
+ * An option 'S' has been added to MixMonitor. If used in combination with
+ the r() and/or t() options, if a frame is available to write to one of
+ those files but not the other, a frame of silence if written to the file
+ that does not have an audio frame. This should prevent the two files
+ from "drifting" when mixed after the fact.
+
+ * If the 'filename' argument to MixMonitor() ended with '.wav49,'
+ Asterisk would silently convert the extension to '.WAV' when opening
+ the file for writing. This caused the MIXMONITOR_FILENAME variable to
+ reference the wrong file. The MIXMONITOR_FILENAME variable will now
+ reflect the name of the file that Asterisk actually used instead of
+ the filename that was passed to the application.
+
+app_page
+------------------
+ * The Page application now tolerates empty positions in the supplied
+ destination list. Dialplan can now be simplified by not having to check
+ for empty positions in the destination list.
+
+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.
+
+chan_pjsip
+------------------
+ * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This
+ allows the behaviour of the moh_passthrough endpoint option to be read or changed
+ in the dialplan. This allows control on a per-call basis.
+
+chan_rtp
+------------------
+ * The UnicastRTP channel driver provided by chan_rtp now accepts
+ "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
+ The first AAAA (preferred) or A record resolved will be used as the destination.
+ The lookup is synchronous so beware of possible dialplan delays if you specify a
+ hostname.
+
+func_curl
+------------------
+ * A new parameter, httpheader, has been added to CURLOPT function. This parameter
+ allows to set custom http headers for subsequent calls off CURL function.
+ Any setting of headers will replace the default curl headers
+ (e.g. "Content-type: application/x-www-form-urlencoded")
+
+ * A new option, followlocation, can now be enabled with the CURLOPT()
+ dialplan function. Setting this will instruct cURL to follow 3xx
+ redirects, which it does not by default.
+
+func_jitterbuffer
+------------------
+ * The JITTERBUFFER dialplan function now has an option to enable video synchronization
+ support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
+ the video is buffered according to the size of the audio jitterbuffer and is
+ synchronized to the audio.
+
+func_volume
+------------------
+ * Accept decimal number as argument.
+
+http
+------------------
+ * You can now disable the /httpstatus page served by Asterisk's built-in
+ HTTP server by setting 'enable_status' to 'no' in http.conf.
+
+minmemfree
+------------------
+ * The 'minmemfree' configuration option now counts memory allocated to
+ the filesystem cache as "free" because it is memory that is available
+ to the process.
+
+res_ari_channels
+------------------
+ * When creating a channel in ARI using the create call
+ you can now specify dialplan variables to be set as part
+ of the same operation.
+
+res_musiconhold
+------------------
+ * This fix allows a realtime moh class to be unregistered from the command
+ line. This is useful when the contents of a directory referenced by a
+ realtime moh class have changed.
+ The realtime moh class is then reloaded on the next request and uses the
+ new directory contents.
+
+ * A new mode - playlist - has been added to res_musiconhold. This mode allows the
+ user to specify the files (or URLs) to play explicitly by putting them directly
+ in musiconhold.conf.
+
+res_pjsip
+------------------
+ * Added a new PJSIP system setting called disable_rport.
+ Default is no to keep support working as before.
+
+ If it is false (default) it adds the 'rport' parameter in the outgoing request message.
+ If it is true it does not add the 'rport' parameter in the outgoing request message.
+
+ This is a system option, but working as a global option.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * In 'type = identify' sections, the addresses specified for the 'match'
+ clause can now include a port number. For IP addresses, the port is
+ provided by including a colon after the address, followed by the
+ desired port number. If supplied, the netmask should follow the port
+ number. To specify a port for IPv6 addresses, the address itself must
+ be enclosed in brackets to be parsed correctly.
+
+res_pjsip_logger
+------------------
+ * The PJSIP packet logger now has the following CLI commands:
+
+ pjsip set logger pcap <filename>
+
+ When used this will create a pcap file containing the incoming
+ and outgoing SIP packets, in unencrypted form.
+
+ pjsip set logger console <on / off>
+
+ This allows you to toggle logging to console on and off.
+
+ pjsip set logger host <IP/subnet mask> add
+
+ This allows you to add an additional IP address or subnet
+ mask to logging, allowing you to log multiple instead of
+ just a single IP address or all traffic.
+
+ The normal "pjsip set logger host" CLI command has also been
+ expanded to allow subnet masks as well.
+
+res_pjsip_session
+------------------
+ * 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
+ contain an audio stream, and not any additional streams such as
+ video.
+
+ * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
+ have been added to res_pjsip endpoints that specify the preferred order
+ of codecs to use between those received/sent in an SDP offer and those
+ set in the endpoint configuration.
+
+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.
+
+ * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
+ an ACL mechanism.
+
+ As such six now options are now available:
+
+ ice_deny
+ ice_permit
+ 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
+ aliases for {stun,ice}_deny.
+
+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.
+
+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.
+
+ * 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.
+
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 17.0.0 --------------------------
------------------------------------------------------------------------------
===
===========================================================
+------------------------------------------------------------------------------
+--- New functionality introduced in Asterisk 18.0.0 --------------------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * The ast_format_cap_from_stream_topology() function has been renamed
+ to ast_stream_topology_get_formats().
+
+app_bridgeaddchan
+------------------
+ * 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.
+
+app_mixmonitor
+------------------
+ * 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).
+
+app_queue
+------------------
+ * The 'Reason' header in the QueueMemberPause AMI Event has been
+ removed. The 'PausedReason' header should be used instead.
+
+ * 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.
+
+app_voicemail
+------------------
+ * 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.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
+------------------------------------------------------------------------------
+
+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.
+
+ARI
+------------------
+ * 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.
+
+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.
+
+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.
+
+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.
+
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 17.0.0 --------------------------
------------------------------------------------------------------------------
+++ /dev/null
-Subject: Core
-Master-Only: True
-
-The Streams API becomes the home for the core ACN capabilities.
-These include...
-
- * Parsing and formatting of codec negotation preferences.
- * Resolving pending streams and topologies with those configured
- using configured preferences.
- * Utility functions for creating string representations of
- streams, topologies, and negotiation preferences.
-
-For codec negotiation preferences:
- * Added ast_stream_codec_prefs_parse() which takes a string
- representation of codec negotiation preferences, which
- may come from a pjsip endpoint for example, and populates
- a ast_stream_codec_negotiation_prefs structure.
- * Added ast_stream_codec_prefs_to_str() which does the reverse.
- * Added many functions to parse individual parameter name
- and value strings to their respectrive enum values, and the
- reverse.
-
-For streams:
- * Added ast_stream_create_resolved() which takes a "live" stream
- and resolves it with a configured stream and the negotiation
- preferences to create a new stream.
- * Added ast_stream_to_str() which create a string representation
- of a stream suitable for debug or display purposes.
-
-For topology:
- * Added ast_stream_topology_create_resolved() which takes a "live"
- topology and resolves it, stream by stream, with a configured
- topology stream and the negotiation preferences to create a new
- topology.
- * Added ast_stream_topology_to_str() which create a string
- representation of a topology suitable for debug or display
- purposes.
- * Renamed ast_format_caps_from_topology() to
- ast_stream_topology_get_formats() to be more consistent with
- the existing ast_stream_get_formats().
-
-Additional changes:
- * A new function ast_format_cap_append_names() appends the results
- to the ast_str buffer instead of replacing buffer contents.
+++ /dev/null
-Subject: ARI Channels
-
-The Channel resource has a new sub-resource "externalMedia".
-This allows an application to create a channel for the sole purpose
-of exchanging media with an external server. Once created, this
-channel could be placed into a bridge with existing channels to
-allow the external server to inject audio into the bridge or
-receive audio from the bridge.
-See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
-for more information.
\ No newline at end of file
+++ /dev/null
-Subject: AMI
-
-You can now specify an optional 'Content-Type' as an argument for the Asterisk
-SendText manager action.
+++ /dev/null
-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.
+++ /dev/null
-Subject: app_chanisavail
-
-The ChanIsAvail application now tolerates empty positions in the supplied
-device list. Dialplan can now be simplified by not having to check for
-empty positions in the device list.
+++ /dev/null
-Subject: app_confbridge
-
-A new bridge profile option, maximum_sample_rate, has been added which sets
-a maximum sample rate that the bridge will be mixed at. This allows the bridge
-to move below the maximum sample rate as needed but caps it at the maximum.
+++ /dev/null
-Subject: app_dial
-
-The Dial application now tolerates empty positions in the supplied
-destination list. Dialplan can now be simplified by not having to check
-for empty positions in the destination list. If there are no endpoints to
-dial then DIALSTATUS is set to CHANUNAVAIL.
+++ /dev/null
-Subject: app_mixmonitor
-
-If the 'filename' argument to MixMonitor() ended with '.wav49,'
-Asterisk would silently convert the extension to '.WAV' when opening
-the file for writing. This caused the MIXMONITOR_FILENAME variable to
-reference the wrong file. The MIXMONITOR_FILENAME variable will now
-reflect the name of the file that Asterisk actually used instead of
-the filename that was passed to the application.
+++ /dev/null
-Subject: app_page
-
-The Page application now tolerates empty positions in the supplied
-destination list. Dialplan can now be simplified by not having to check
-for empty positions in the destination list.
+++ /dev/null
-Subject: ARI
-
-A new parameter 'inhibitConnectedLineUpdates' is now available in the
-'bridges.addChannel' call. This prevents the identity of the newly connected
-channel from being presented to other bridge members.
+++ /dev/null
-Subject: res_ari_channels
-
-When creating a channel in ARI using the create call
-you can now specify dialplan variables to be set as part
-of the same operation.
+++ /dev/null
-Subject: chan_pjsip
-
-A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This
-allows the behaviour of the moh_passthrough endpoint option to be read or changed
-in the dialplan. This allows control on a per-call basis.
+++ /dev/null
-Subject: chan_rtp
-
-The UnicastRTP channel driver provided by chan_rtp now accepts
-"<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
-The first AAAA (preferred) or A record resolved will be used as the destination.
-The lookup is synchronous so beware of possible dialplan delays if you specify a
-hostname.
+++ /dev/null
-Subject: app_confbridge
-
-A new option, "text_messaging", has been added to the user profile
-which allows control over whether text messaging is enabled or
-disabled for a user. If enabled (the default) text messages
-will be sent to the user. If disabled no text messages will be
-sent to the user.
+++ /dev/null
-Subject: Features
-
-Adds support for AudioSocket, a very simple bidirectional audio streaming
-protocol. There are both channel and application interfaces.
-
-A description of the protocol can be found on the referenced wiki page. A
-short talk about the reasons and implementation can be found on YouTube at
-the link provided.
-
-ARI support has also been added via the existing "externalMedia" ARI
-functionality. The UUID is specified using the arbitrary "data" field.
-
-Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
-YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
+++ /dev/null
-Subject: func_curl
-
-A new option, followlocation, can now be enabled with the CURLOPT()
-dialplan function. Setting this will instruct cURL to follow 3xx
-redirects, which it does not by default.
+++ /dev/null
-Subject: func_curl
-
-A new parameter, httpheader, has been added to CURLOPT function. This parameter
-allows to set custom http headers for subsequent calls off CURL function.
-Any setting of headers will replace the default curl headers
-(e.g. "Content-type: application/x-www-form-urlencoded")
+++ /dev/null
-Subject: func_jitterbuffer
-
-The JITTERBUFFER dialplan function now has an option to enable video synchronization
-support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
-the video is buffered according to the size of the audio jitterbuffer and is
-synchronized to the audio.
+++ /dev/null
-Subject: func_volume
-
-Accept decimal number as argument.
+++ /dev/null
-Subject: Core
-
-H.265/HEVC is now a supported video codec and it can be used by
-specifying "h265" in the allow line.
-Please note however, that handling of the additional SDP parameters
-described in RFC 7798 section 7.2 is not yet supported.
+++ /dev/null
-Subject: Messaging
-
-In order to reduce the amount of AMI and ARI events generated,
-the global "Message/ast_msg_queue" channel can be set to suppress
-it's normal channel housekeeping events such as "Newexten",
-"VarSet", etc. This can greatly reduce load on the manager
-and ARI applications when the Digium Phone Module for Asterisk
-is in use. To enable, set "hide_messaging_ami_events" in
-asterisk.conf to "yes" In Asterisk versions <18, the default
-is "no" preserving existing behavior. Beginning with
-Asterisk 18, the option will default to "yes".
+++ /dev/null
-Subject: http
-
-You can now disable the /httpstatus page served by Asterisk's built-in
-HTTP server by setting 'enable_status' to 'no' in http.conf.
+++ /dev/null
-Subject: minmemfree
-
-The 'minmemfree' configuration option now counts memory allocated to
-the filesystem cache as "free" because it is memory that is available
-to the process.
+++ /dev/null
-Subject: app_mixmonitor
-
-An option 'S' has been added to MixMonitor. If used in combination with
-the r() and/or t() options, if a frame is available to write to one of
-those files but not the other, a frame of silence if written to the file
-that does not have an audio frame. This should prevent the two files
-from "drifting" when mixed after the fact.
+++ /dev/null
-Subject: res_musiconhold
-
-A new mode - playlist - has been added to res_musiconhold. This mode allows the
-user to specify the files (or URLs) to play explicitly by putting them directly
-in musiconhold.conf.
+++ /dev/null
-Subject: res_pjsip_logger
-
-The PJSIP packet logger now has the following CLI commands:
-
-pjsip set logger pcap <filename>
-
-When used this will create a pcap file containing the incoming
-and outgoing SIP packets, in unencrypted form.
-
-pjsip set logger console <on / off>
-
-This allows you to toggle logging to console on and off.
-
-pjsip set logger host <IP/subnet mask> add
-
-This allows you to add an additional IP address or subnet
-mask to logging, allowing you to log multiple instead of
-just a single IP address or all traffic.
-
-The normal "pjsip set logger host" CLI command has also been
-expanded to allow subnet masks as well.
+++ /dev/null
-Subject: res_musiconhold
-
-This fix allows a realtime moh class to be unregistered from the command
-line. This is useful when the contents of a directory referenced by a
-realtime moh class have changed.
-The realtime moh class is then reloaded on the next request and uses the
-new directory contents.
+++ /dev/null
-Subject: res_pjsip
-
-Added a new PJSIP system setting called disable_rport.
-Default is no to keep support working as before.
-
-If it is false (default) it adds the 'rport' parameter in the outgoing request message.
-If it is true it does not add the 'rport' parameter in the outgoing request message.
-
-This is a system option, but working as a global option.
\ No newline at end of file
+++ /dev/null
-Subject: res_pjsip
-Subject: res_pjsip_session
-Master-Only: True
-
-Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
-have been added to res_pjsip endpoints that specify the preferred order
-of codecs to use between those received/sent in an SDP offer and those
-set in the endpoint configuration.
+++ /dev/null
-Subject: res_pjsip_endpoint_identifier_ip
-
-In 'type = identify' sections, the addresses specified for the 'match'
-clause can now include a port number. For IP addresses, the port is
-provided by including a colon after the address, followed by the
-desired port number. If supplied, the netmask should follow the port
-number. To specify a port for IPv6 addresses, the address itself must
-be enclosed in brackets to be parsed correctly.
+++ /dev/null
-Subject: res_pjsip_session
-
-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
-contain an audio stream, and not any additional streams such as
-video.
+++ /dev/null
-Subject: res_rtp_asterisk
-
-The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
-an ACL mechanism.
-
-As such six now options are now available:
-
-ice_deny
-ice_permit
-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
-aliases for {stun,ice}_deny.
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-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'.
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-Subject: Core
-Master-Only: True
-
-The ast_format_cap_from_stream_topology() function has been renamed
-to ast_stream_topology_get_formats().
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-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).
+++ /dev/null
-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.
+++ /dev/null
-Subject: app_queue
-Master-Only: True
-
-The 'Reason' header in the QueueMemberPause AMI Event has been
-removed. The 'PausedReason' header should be used instead.
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-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.
+++ /dev/null
-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.
projects / thirdparty / asterisk.git / commitdiff
