+++ /dev/null
-## **DO NOT REMOVE THIS FILE!**
-
-The only files that should be added to this directory are ones that will be
-used by the release script to update the CHANGES file automatically. The only
-time that it is necessary to add something to the CHANGES-staging directory is
-if you are either adding a new feature to Asterisk or adding new functionality
-to an existing feature. The file does not need to have a meaningful name, but
-it probably should. If there are multiple items that need documenting, you can
-add multiple files, each with their own description. If the message is going to
-be the same for each subject, then you can add multiple subject headers to one
-file. The "Subject: xxx" line is case sensitive! For example, if you are making
-a change to PJSIP, then you might add the file "res_pjsip_my_cool_feature.txt" to
-this directory, with a short description of what it does. The files must have
-the ".txt" suffix. If you are adding multiple entries, they should be done in
-the same commit to avoid merge conflicts. Here's an example:
-
-> Subject: res_pjsip
-> Subject: Core
->
-> Here's a pretty good description of my new feature that explains exactly what
-> it does and how to use it.
-
-Here's a master-only example:
-
-> Subject: res_ari
-> Master-Only: True
->
-> This change will only go into the master branch. The "Master-Only" header
-> will never be in a change not in master.
-
-Note that the second subject has another header: "Master-Only". Changes that go
-into the master branch and ONLY the master branch are the only ones that should
-have this header. Also, the value can only be "true" or "True". The
-"Master-Only" part of the header IS case-sensitive, however!
-
-For more information, check out the wiki page:
-https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
+++ /dev/null
-Subject: app_senddtmf
-
-The SendFlash AMI action now allows sending
-a hook flash event on a channel.
+++ /dev/null
-Subject: pbx_builtins
-
-It is now possible to not wait for media on
-a channel when answering it using Answer,
-by specifying the i option.
+++ /dev/null
-Subject: app_amd
-
-An audio file to play during AMD processing can
-now be specified to the AMD application or configured
-in the amd.conf configuration file.
+++ /dev/null
-Subject: app_bridgewait
-
-Adds the n option to not answer the channel when
-the BridgeWait application is called.
+++ /dev/null
-Subject: app_broadcast
-
-A Broadcast application is now available which allows
-for asynchronous one-to-many and many-to-one channel audio.
+++ /dev/null
-Subject: app_confbridge
-
-Adds the end_marked_any option which can be used
-to kick users from a conference after any
-marked user leaves (including marked users).
+++ /dev/null
-Subject: app_directory
-
-A new option 's' has been added to the Directory() application that
-will skip calling the extension and instead set the extension as
-DIRECTORY_EXTEN channel variable.
+++ /dev/null
-Subject: app_if
-
-Adds the If, ElseIf, Else, EndIf, and ExitIf applications
-for conditional execution of a block of code.
+++ /dev/null
-Subject: app_mixmonitor
-
-Adds the c option to use the real Caller ID on
-the channel in voicemail recordings as opposed
-to the Connected Line.
+++ /dev/null
-Subject: app_mixmonitor
-
-The d option for MixMonitor now allows deleting
-the original recording when MixMonitor exits,
-which can be useful when MixMonitor copies it
-somewhere else before exiting.
+++ /dev/null
-Subject: app_mixmonitor
-Subject: audiohook
-Subject: manager
-
-It is now possible to specify the MixMonitorID when calling
-the manager action: MixMonitorMute. This will allow an
-individual MixMonitor instance to be muted via ID.
-
-The MixMonitorID can be stored as a channel variable using
-the 'i' MixMonitor option and is returned upon creation if
-this option is used.
-
-As part of this change, if no MixMonitorID is specified in
-the manager action MixMonitorMute, Asterisk will set the mute
-flag on all MixMonitor audiohooks on the channel. Previous
-behavior would set the flag on the first MixMonitor audiohook
-found.
+++ /dev/null
-Subject: app_read
-
-A new option 'e' has been added to allow Read() to return the
-terminator as the dialed digits in the case where only the terminator
-is entered.
+++ /dev/null
-Subject: app_senddtmf
-
-A new option has been added to SendDTMF() which will answer the
-specified channel if it is not already up. If no channel is specified,
-the current channel will be answered instead.
+++ /dev/null
-Subject: app_signal
-
-Adds Signal and WaitForSignal applications
-which can be used for signaling or as a
-simple message queue in the dialplan.
+++ /dev/null
-Subject: app_voicemail
-
-The voicemail user option attachextrecs can
-now be set to control whether external recordings
-trigger voicemail email notifications.
+++ /dev/null
-Subject: bridge_builtin_features
-
-Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-
-Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
-interval in seconds will result in a periodic beep being
-played to the monitored channel upon MixMontior/Monitor
-feature start.
-
-If an interval less than 5 seconds is specified, the interval
-will default to 5 seconds. If the value is set to an invalid
-interval, the default of 15 seconds will be used.
+++ /dev/null
-Subject: cdr
-
-Two new options have been added which allow
-bridging and dial state changes to be ignored
-in CDRs, which can be useful if a single CDR
-is desired for a channel.
+++ /dev/null
-Subject: cli
-Subject: core
-
-This change increases the display width on 'core show channels'
-amd 'core show channels verbose'
-
-For 'core show channels', the Channel name field is increased to
-64 characters and the Location name field is increased to 32
-characters.
-
-For 'core show channels verbose', the Channel name field is
-increased to 80 characters, the Context is increased to 24
-characters and the Extension is increased to 24 characters.
-
+++ /dev/null
-Subject: db
-
-The DBPrefixGet AMI action now allows retrieving
-all of the DB keys beginning with a particular
-prefix.
+++ /dev/null
-Subject: DUNDi
-
-DUNDi now supports chan_pjsip. Outgoing calls using
-PJSIP require the pjsip_outgoing_endpoint option
-to be set in dundi.conf.
+++ /dev/null
-Subject: features
-
-The Bridge application now has the n "no answer" option
-that can be used to prevent the channel from being
-automatically answered prior to bridging.
+++ /dev/null
-Subject: format_sln
-
-format_sln now recognizes '.slin' as a valid
-file extension in addition to the existing
-'.sln' and '.raw'.
+++ /dev/null
-Subject: New EXPORT function
-
-A new function, EXPORT, allows writing variables
-and functions on other channels, the complement
-of the IMPORT function.
+++ /dev/null
-Subject: func_json
-
-Additional parsing capabilities have been added to the
-JSON_DECODE function, including support for arrays
-and recursive indexing.
+++ /dev/null
-Subject: func_strings
-
-Three new functions, TRIM, LTRIM, and RTRIM, are
-now available for trimming leading and trailing
-whitespace.
+++ /dev/null
-Subject: chan_dahdi
-
-FXO channels (FXS signaled) that don't use callerid or
-distinctive ring detection can now be configured
-to enter the dialplan immediately using immediate=yes,
-instead of waiting for at least one ring.
+++ /dev/null
-Subject: http
-Master-Only: True
-
-For bound addresses, the HTTP status page now combines the bound
-address and bound port in a single line. Additionally, the SSL bind
-address has been renamed to TLS.
+++ /dev/null
-Subject: locks
-
-A new AMI event, DeadlockStart, is now available
-when Asterisk is compiled with DETECT_DEADLOCKS,
-and can indicate that a deadlock has occured.
+++ /dev/null
-Subject: AMI
-
-The AOCMessage action can now be used to generate AOC-S messages.
+++ /dev/null
-Subject: res_geolocation
-
-* Added processing for the 'confidence' element.
-* Added documentation to some APIs.
-* removed a lot of complex code related to the very-off-nominal
- case of needing to process multiple location info sources.
-* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
- one eprofile instead of a datastore of multiples.
-* Plugged a huge leak in XML processing that arose from
- insufficient documentation by the libxml/libxslt authors.
-* Refactored stylesheets to be more efficient.
-* Renamed 'profile_action' to 'profile_precedence' to better
- reflect it's purpose.
-* Added the config option for 'allow_routing_use' which
- sets the value of the 'Geolocation-Routing' header.
-* Removed the GeolocProfileCreate and GeolocProfileDelete
- dialplan apps.
-* Changed the GEOLOC_PROFILE dialplan function as follows:
- * Removed the 'profile' argument.
- * Automatically create a profile if it doesn't exist.
- * Delete a profile if 'inheritable' is set to no.
-* Fixed various bugs and leaks
-* Updated Asterisk WiKi documentation.
-
-Added 4 built-in profiles:
- "<prefer_config>"
- "<discard_config>"
- "<prefer_incoming>"
- "<discard_incoming>"
-The profiles are empty except for having their precedence
-set.
-
-Added profile parameter "suppress_empty_ca_elements" that
-will cause Civic Address elements that are empty to be
-suppressed from the outgoing PIDF-LO document.
-
-You can now specify the location object's format, location_info,
-method, location_source and confidence parameters directly on
-a profile object for simple scenarios where the location
-information isn't common with any other profiles. This is
-mutually exclusive with setting location_reference on the
-profile.
-
-Added an 'a' option to the GEOLOC_PROFILE function to allow
-variable lists like location_info_refinement to be appended
-to instead of replacing the entire list.
-
-Added an 'r' option to the GEOLOC_PROFILE function to resolve all
-variables before a read operation and after a Set operation.
+++ /dev/null
-Subject: Add support for named capture agent.
-
-A name for the capture agent can now be specified
-using the capture_name option which, if specified,
-will be sent to the HEP server.
+++ /dev/null
-Subject: res_http_media_cache
-
-The res_http_media_cache module now attempts to load
-configuration from the res_http_media_cache.conf file.
-The following options were added:
- * timeout_secs
- * user_agent
- * follow_location
- * max_redirects
- * protocols
- * redirect_protocols
- * dns_cache_timeout_secs
+++ /dev/null
-Subject: res_musiconhold_answeredonly
-
-This change adds an option, answeredonly, that will prevent music
-on hold on channels that are not answered.
+++ /dev/null
-Subject: res_phoneprov
-
-On multihomed Asterisk servers with dynamic SERVER template variables,
-reloading this module is no longer required when re-provisioning your
-phone to another interface address (e.g. when moving between VLANs.)
+++ /dev/null
-Subject: res_pjsip
-
-A new option named "peer_supported" has been added to the endpoint option
-100rel. When set to this option, Asterisk sends provisional responses
-reliably if the peer supports it. If the peer does not support reliable
-provisional responses, Asterisk sends them normally.
+++ /dev/null
-Subject: res_pjsip
-
-A new option named "all_codecs_on_empty_reinvite" has been added to the
-global section. When this option is enabled, on reception of a re-INVITE
-without SDP, Asterisk will send an SDP offer in the 200 OK response containing
-all configured codecs on the endpoint, instead of simply those that have
-already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
-The default value is "off".
\ No newline at end of file
+++ /dev/null
-Subject: res_pjsip_aoc
-
-Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
-A new endpoint option, send_aoc, controls this.
+++ /dev/null
-Subject: res_pjsip_logger
-
-SIP messages can now be filtered by SIP request method
-(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
-SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
-allowing for more granular debugging to be done
-in the CLI. This applies to requests but not responses.
+++ /dev/null
-Subject: res_pjsip_notify
-
-Allows using the config options in pjsip_notify.conf
-from AMI actions as with the existing CLI commands.
+++ /dev/null
-Subject: res_pjsip_header_funcs
-
-The new PJSIP_HEADER_PARAM function now fully supports both
-URI and header parameters. Both reading and writing
-parameters are supported.
+++ /dev/null
-Subject: res_pjsip
-
-Added options "security_negotiation" and "security_mechanisms" to pjsip
-endpoints and registrations. "security_negotiation" can be set to "no" (default)
-or "mediasec", and "security_mechanisms" can be a list of comma-separated
-security_mechanisms in the form defined by RFC 3329 section 2.2.
+++ /dev/null
-Subject: res_pjsip_session
-
-The overlap_context option now allows explicitly
-specifying a context to use for overlap dialing matches.
+++ /dev/null
-Subject: res_pjsip
-
-TLS transports in res_pjsip can now reload their TLS certificate
-and private key files, provided the filename of them has not
-changed.
+++ /dev/null
-subject: res_pjsip
-
-user_eq_phone=yes flag on a pjsip endpoint will now set user=phone on
-the From and Prviacy headers in addition to the existing To and RURI
+++ /dev/null
-Subject: res_rtp_asterisk
-
-This module has been updated to provide additional
-quality statistics in the form of an Asterisk
-Media Experience Score. The score is available using
-the same mechanisms you'd use to retrieve jitter, loss,
-and rtt statistics. For more information about the
-score and how to retrieve it, see
-https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
+++ /dev/null
-Subject: res_pjsip_rfc3326
-
-Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
-addition to currently supported Q.850). The first header found will be used to set
-the HANGUPCAUSE variable.
+++ /dev/null
-Subject: res_tonedetect
-
-The TONE_DETECT function now supports
-detection of audible ringback tone
-using the p option.
+++ /dev/null
-Subject: test.c
-
-The "tests" attribute of the "testsuite" element in the
-output XML now reflects only the tests actually requested
-to be executed instead of all the tests registered.
-
-The "failures" attribute was added to the "testsuite"
-element.
-
-Also added two new unit tests that just pass and fail
-to be used for testing CI itself.
+++ /dev/null
-Subject: Transfer feature
-
-The following capabilities have been added to the
-transfer feature:
-
-- The transfer initiation announcement prompt can
-now be customized in features.conf.
-
-- The TRANSFER_EXTEN variable now can be set on the
-transferer's channel in order to allow the transfer
-function to automatically attempt to go to the extension
-contained in this variable, if it exists. The transfer
-context behavior is not changed (TRANSFER_CONTEXT is used
-if it exists; otherwise the default context is used).
+++ /dev/null
-Subject: xmldocs
-
-The XML documentation can now be reloaded without restarting
-Asterisk, which makes it possible to load new modules that
-enforce documentation without restarting Asterisk.
+++ /dev/null
-## **DO NOT REMOVE THIS FILE!**
-
-The only files that should be added to this directory are ones that will be
-used by the release script to update the UPGRADE.txt file automatically. The
-only time that it is necessary to add something to the UPGRADE-staging directory
-is if you are making a breaking change to an existing feature in Asterisk. The
-file does not need to have a meaningful name, but it probably should. If there
-are multiple items that need documenting, you can add multiple files, each with
-their own description. If the message is going to be the same for each subject,
-then you can add multiple subject headers to one file. The "Subject: xxx" line
-is case sensitive! For example, if you are making a change to PJSIP, then you
-might add the file "res_pjsip_my_cool_feature.txt" to this directory, with a
-short description of what it does. The files must have the ".txt" suffix.
-If you are adding multiple entries, they should be done in the same commit
-to avoid merge conflicts. Here's an example:
-
-> Subject: res_pjsip
-> Subject: Core
->
-> Here's a pretty good description of my new feature that explains exactly what
-> it does and how to use it.
-
-Here's a master-only example:
-
-> Subject: res_ari
-> Master-Only: True
->
-> This change will only go into the master branch. The "Master-Only" header
-> will never be in a change not in master.
-
-Note that the second subject has another header: "Master-Only". Changes that go
-into the master branch and ONLY the master branch are the only ones that should
-have this header. Also, the value can only be "true" or "True". The
-"Master-Only" part of the header IS case-sensitive, however!
-
-For more information, check out the wiki page:
-https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
+++ /dev/null
-Subject: app_cdr
-Master-Only: True
-
-The previously deprecated NoCDR application has been removed.
-Additionally, the previously deprecated 'e' option to the ResetCDR
-application has been removed.
+++ /dev/null
-Subject: app_macro
-Master-Only: True
-
-This module was deprecated in Asterisk 16
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
-
-
-For most modules that interacted with app_macro,
-this change is limited to no longer looking for
-the current context from the macrocontext when set.
-
-The following modules have additional impacts:
-
-app_dial - no longer supports M^ connected/redirecting macro
-
-app_minivm - samples written using macro will no longer work.
-The sample needs to be re-written
-
-app_queue - can no longer call a macro on the called party's
-channel. Use gosub which is currently supported
-
-ccss - no callback macro, gosub only
-
-app_voicemail - no macro support
-
-channel - remove macrocontext and priority, no connected
-line or redirection macro options
-
-options - stdexten is deprecated to gosub as the default
-and only options
-
-pbx - removed macrolock
-
-pbx_dundi - no longer look for macro
-
-snmp - removed macro context, exten, and priority
+++ /dev/null
-Subject: app_osplookup
-Master-Only: True
-
-This module was deprecated in Asterisk 19
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
+++ /dev/null
-Subject: app_playback
-
-In Asterisk 11, if a channel was redirected away during Playback(),
-the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
-(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
-behavior was inadvertently changed and the same operation would result
-in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
-behavior has been restored.
+++ /dev/null
-Subject: chan_alsa
-Master-Only: True
-
-This module was deprecated in Asterisk 19
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
+++ /dev/null
-Subject: chan_mgcp
-Master-Only: True
-
-This module was deprecated in Asterisk 19
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
-
+++ /dev/null
-Subject: chan_sip
-Master-Only: True
-
-This module was deprecated in Asterisk 17
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
+++ /dev/null
-Subject: chan_skinny
-Master-Only: True
-
-This module was deprecated in Asterisk 19
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
+++ /dev/null
-Subject: AMI (Asterisk Manager Interface)
-
-Previously, GetConfig and UpdateConfig were able to access files outside of
-the Asterisk configuration directory. Now this access is put behind the
-live_dangerously configuration option in asterisk.conf, which is disabled by
-default. If access to configuration files outside of the Asterisk configuation
-directory is required via AMI, then the live_dangerously configuration option
-must be set to yes.
+++ /dev/null
-Subject: pbx_builtins
-Master-Only: True
-
-The previously deprecated ImportVar and SetAMAFlags
-applications have now been removed.
+++ /dev/null
-Subject: res_crypto
-
-In addition to only paying attention to files ending with .key or .pub
-in the keys directory, we now also ignore any files which aren't regular
-files.
+++ /dev/null
-Subject: res_monitor
-Master-Only: True
-
-This module was deprecated in Asterisk 16
-and is now being removed in accordance with
-the Asterisk Module Deprecation policy.
-
-This also removes the 'w' and 'W' options
-for app_queue.
-
-MixMonitor should be default and only option
-for all settings that previously used either
-Monitor or MixMonitor.
+++ /dev/null
-Subject: translate.c
-Master-Only: True
-
-When setting up translation between two codecs the quality was not taken into account,
-resulting in suboptimal translation. The quality is now taken into account,
-which can reduce the number of translation steps required, and improve the resulting quality.
- \ref manager.c Main manager code file
*/
-#define AMI_VERSION "10.0.0"
+#define AMI_VERSION "11.0.0"
#define DEFAULT_MANAGER_PORT 5038 /* Default port for Asterisk management via TCP */
#define DEFAULT_MANAGER_TLS_PORT 5039 /* Default port for Asterisk management via TCP */
const char *from;
/*! The body of the message */
const char *body;
- /*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */
struct ast_json *variables;
};
/*!
const char *from;
/*! The body of the message */
const char *body;
- /*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */
struct ast_json *variables;
};
/*!
"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
"_author": "David M. Lee, II <dlee@digium.com>",
"_svn_revision": "$Revision$",
- "apiVersion": "9.0.0",
+ "apiVersion": "10.0.0",
"swaggerVersion": "1.1",
"basePath": "http://localhost:8088/ari",
"apis": [
projects / thirdparty / asterisk.git / commitdiff
