MusicOnHold

• MusicOnHold streams (all modes other than "files") now support wide band audio.

MixMonitor

• A new option, B(), has been added that will turn on a periodic beep while the call is being recorded.
• New options to play a beep when starting a recording and stopping a recording have been added. The option 'p' will play a beep to the channel that starts the recording. The option 'P' will play a beep to the channel that stops the recording.

Monitor

• A new option, B(), has been added that will turn on a periodic beep while the call is being recorded.

Page

• Added options 'b' and 'B' to apply pre-dial handlers for outgoing calls and for the channel executing Page respectively.

PickupChan

• PickupChan now accepts channel uniqueids of channels to pickup.

ReadFile

• The module app_readfile was deprecated and has been removed. Users of ReadFile should use func_env's FILE function instead.

Record

• The Record application now has an option 'o' which allows 0 to act as an exit key. This will set the the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'.

Say

• If the channel variable SAY_DTMF_INTERRUPT is present on a channel and set to 'true' (case insensitive), then any Say application (SayNumber, SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will anticipate DTMF. If DTMF is received, these applications will behave like the background application and jump to the received extension once a match is established or after a short period of inactivity.
• The Say family of dialplan applications now support the Japanese language. The language parameter in say.conf now recognizes a setting of ja, which will enable Japanese language specific mechanisms for playing back numbers, dates, and other items.

SayCountPL

• The module app_saycountpl was deprecated and has been removed. Users of app_saycountpl should use the Say family of applications.

SetMusicOnHold

• The SetMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use the CHANNEL function's musicclass setting instead.

VoiceMail

• VoiceMail and VoiceMailMain now support the Japanese language. The language parameter in voicemail.conf now recognizes a setting of ja, which will enable prompts to be played back using a Japanese grammatical structure. Additional prompts are necessary for this functionality, including:
  • jb-arimasu: there is
  • jb-arimasen: there is not
  • jb-oshitekudasai: please press
  • jb-ni: article ni
  • jb-ga: article ga
  • jb-wa: article wa
  • jb-wo: article wo
• VoiceMail mailboxes configured in voicemail.conf can now have multiple e-mail address specified for a single mailbox. Each e-mail address is separated by the | character.

WaitMusicOnHold

• The WaitMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use MusicOnHold with a duration parameter instead.

Build System

• The location of the sample configuration files delivered with Asterisk have been moved from configs to configs/samples. This allows for other sample configuration sets to be defined in the future. The action of make samples is exactly the same as previous versions of Asterisk.
• The menuselect tool has been pulled into the Asterisk repository. Generally, this change is transparent to those using tarballs of Asterisk; to those working directly with the Asterisk repository, there is no accessing of the menuselect or mxml external repositories.
• The menuselect tool no longer uses a bundled mxml library. Instead, it now uses libxml2. As a result,  the libxml2 development library is now a required dependency for Asterisk.

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  res_mwi_external_ami res_mwi_external_ami

  Added a new module that provides AMI control over MWI within Asterisk, res_mwi_external_ami. Note that this module depends on res_mwi_external; for more information on enabling this module, see res_mwi_external. This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as the MWIGet/MWIGetComplete events.

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• New DeviceStateChanged and PresenceStateChanged AMI events have been added. These events are emitted whenever a device state or presence state change occurs. The events are controlled by res_manager_device_state.so and res_manager_presence_state.so. If the high frequency of these events is problematic for you, do not load these modules.
• New events have been added for the TALK_DETECT function. When the function is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be emitted to connected AMI clients indicating the start/stop of talking on the channel.
• The DialStatus field in the DialEnd event can now contain additional statuses that convey how the dial operation terminated. This includes ABORT, CONTINUE, and GOTO.

...

• Operations that create a resource can now provide the unique identifier as a parameter to the creation request. This includes:
  • Channels:
    • A channelId can now be provided when creating a channel, either in the request URI (POST channels/my-channel-id) or as a query parameter. A Local channel will suffix the second channel id with ;2 unless the otherChannelId is provided as a query parameter.
    • A snoop channel can be started with a snoopId, in the request URI (POST channels/my-channel-id/snoop/my-snoop-id) or as a query parameter.
  • Bridges: A bridgeId can now be provided when creating a bridge, either in the request URI (POST bridges/my-bridge-id) or as a query parameter.
  • Playbacks: A playbackId can be provided when starting a playback, either in the request URI (POST channels/my-channel-id/play/my-playback-id or POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
• Bridges: the bridge type used when creating a bridge is now a comma separated list of bridge properties. Valid options are: mixing, holding, dtmf_events, and proxy_media.
• The LiveRecording object in recording events now contains a target_uri field which contains the URI of what is being recorded.
• Stored recordings now support a new operation, copy. This will take an existing stored recording and copy it to a new location in the recordings directory.
• LiveRecording objects now have three additional fields that can be reported in a RecordingFinished ARI event:
  • total_duration: the duration of the recording.
  • talking_duration: optional. The duration of talking detected in the recording. This is only available if max_silence_seconds was specified when the recording was started.
  • silence_duration: optional. The duration of silence detected in the recording. This is only available if max_silence_seconds was specified when the recording was started.
  Note that all duration values are reported in seconds.

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  ari_text_message ari_text_message

  Users of ARI can now send and receive out of call text messages. Messages can be sent using a sendMessage operation either directly to a particular endpoint or to the endpoints resource directly. In the latter case, the destination is derived from the URI scheme. Text messages are passed to ARI clients as TextMessageReceived events. ARI clients can choose to receive text messages by subscribing to the particular endpoint technology or endpoints that they are interested in.
• The applications resource now supports subscriptions to all endpoints of a particular channel technology. For example, subscribing to an eventSource of endpoint:PJSIP will subscribe to all PJSIP endpoints.
• New event models have been added for the  TALK_DETECT function. When the function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished events will be emitted to connected WebSockets subscribed to the channel, indicating the start/stop of talking on the channel.
• A new Playback URI tone has been added. Tones are specified either as an indication name, e.g., tone:busy , from indications.conf or as a tone pattern, e.g., tone:240/250,0/250. Tones differ from normal playback URIs in that they must be stopped manually and will continue to occupy a channel's ARI control queue until they are stopped. They also can not be rewound or fast-forwarded.
• User events can now be generated from ARI. Events can be signalled with arbitrary JSON variables, and include one or more of channel, bridge, or endpoint snapshots.  An application must be specified which will receive the event message (other applications can subscribe to it). If a channel is specified, the message will also be delivered to connected AMI clients. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a Stasis application they are attached to or if something is subscribed to the channel.
• The Bridge data model now contains the additional fields name and creator. The name field conveys a descriptive name for the bridge; the creator field conveys the name of the entity that created the bridge. This affects all responses to HTTP requests that return a Bridge data model as well as all event derived data models that contain a Bridge data model. The POST /bridges operation may now optionally specify a name to give to the bridge being created.

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  res_ari_mailboxes res_ari_mailboxes

  Added a new ARI resource mailboxes which allows the creation and modification of mailboxes managed by external MWI. Modules res_mwi_external and res_stasis_mailbox must be enabled to use this resource. For more information on external MWI control, see res_mwi_external.
• Added new events for externally initiated transfers. The event BridgeBlindTransfer is now raised when a channel initiates a blind transfer of a bridge in the ARI controlled application to the dialplan; the BridgeAttendedTransfer event is raised when a channel initiates an attended transfer of a bridge in the ARI controlled application to the dialplan.
• Channel variables may now be specified as a body parameter to the POST /channels operation. The variables key in the JSON is interpreted as a sequence of key/value pairs that will be added to the created channel as channel variables. Other parameters in the JSON body are treated as query parameters of the same name.

...

• A new module, res_mwi_external, has been added to Asterisk. This module acts as a base framework that other modules can build on top of to allow an external system to control MWI within Asterisk. For implementations that make use of res_mwi_external, see the res_mwi_external_ami notes under the AMI changes and res_ari_mailboxes notes under the ARI changes. Note that res_mwi_external conflicts with other modules that may produce MWI themselves, such as app_voicemail. res_mwi_external and other modules that depend on it cannot be built or loaded with app_voicemail present.

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• A new module, res_pjsip_multihomed handles situations where the system Asterisk is running out has multiple interfaces. res_pjsip_multihomed determines which interface should be used during message sending.

res_pjsip_outbound_publish

• A new module, res_pjsip_outbound_publish provides the mechanisms for sending PUBLISH requests for specific event packages to another SIP User Agent. See Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_outbound_registration

• A new CLI command has been added: pjsip show registrations, which lists all configured PJSIP registrations.

res_pjsip_pidf_digium_body_supplement

• A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY request body formatting for presence support in Digium phones.

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res_pjsip_pubsub res_pjsip_pubsub

res_pjsip_pubsub

• Subscriptions can now be persisted via the subscription_persistence object in pjsip.conf. Note that it is up to the configuration in sorcery.conf to determine how the subscription is persisted.
• The publish/subscribe core module has been updated to support RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS). Resource lists are configured in pjsip.conf under a new object type, resource_list. Resource lists can contain either message-summary or presence events, can be composed of specific resources that provide the event, or other resource lists.
• Inbound publication support is provided by a new object, inbound-publication. This configures res_pjsip_pubsub to accept PUBLISH requests from a particular resource. Which events are accepted is constructed dynamically; see res_pjsip_publish_asterisk for more information and Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_pidf_digium_body_supplement

• A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY request body formatting for presence support in Digium phones.

res_pjsip_send_to_voicemail

• A new module, res_pjsip_send_to_voicemail allows for REFER requests with particular headers to transfer a PJSIP channel directly to a particular extension that has VoiceMail. This is intended to be used with Digium phones that support this feature.

res_pjsip_outbound_publish

• A new module, res_pjsip_outbound_publish provides the mechanisms for sending PUBLISH requests for specific event packages to another SIP User Agent. See Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_outbound_registration

• A new CLI command has been added: pjsip show registrations, which lists all configured PJSIP registrations.

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res_pjsip_publish_asterisk res_pjsip_publish_asterisk

res_pjsip_publish_asterisk

• A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of Asterisk information to other Asterisk servers. This module is intended only for Asterisk to Asterisk exchanges of information. Currently, this includes both mailbox state and device state information. See Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_send_to_voicemail

• A new module, res_pjsip_send_to_voicemail allows for REFER requests with particular headers to transfer a PJSIP channel directly to a particular extension that has VoiceMail. This is intended to be used with Digium phones that support this feature.