1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
13 ------------------------------------------------------------------------------
17 * This module was deprecated and has been removed. Users of app_dahdibarge
18 should use ChanSpy instead.
22 * This module was deprecated and has been removed. Users of app_readfile
23 should use func_env's FILE function instead.
27 * This module was deprecated and has been removed. Users of app_saycountpl
28 should use the Say family of applications.
32 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
33 These events are emitted whenever a device state or presence state change
34 occurs. The events are controlled by res_manager_device_state.so and
35 res_manager_presence_state.so. If the high frequency of these events is
36 problematic for you, do not load these modules.
38 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
39 work in basically the same way as the 'dialplan add extension' and
40 'dialplan remove extension' CLI commands respectively.
42 * New AMI action LoggerRotate reloads and rotates logger in the same manner
43 as CLI command 'logger rotate'
45 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
46 functionality of CLI commands 'fax show sessions', 'fax show session',
47 and fax show stats' respectively.
49 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
50 enable manager control over PRI debugging levels and file output.
54 * This module was deprecated and has been removed. Users of cdr_sqlite
55 should use cdr_sqlite3_custom.
59 * Added the ability to support PostgreSQL application_name on connections.
60 This allows PostgreSQL to display the configured name in the
61 pg_stat_activity view and CSV log entries. This setting is configurable
62 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
66 * Added the ability to support PostgreSQL application_name on connections.
67 This allows PostgreSQL to display the configured name in the
68 pg_stat_activity view and CSV log entries. This setting is configurable
69 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
73 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
74 and BRIDGE_EXIT events.
78 * SS7 support now requires libss7 v2.0 or later.
80 * Added SS7 support for connected line and redirecting.
82 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
85 * Added several SS7 config option parameters described in
86 chan_dahdi.conf.sample.
90 * This module was deprecated and has been removed. Users of chan_gtalk
91 should use chan_motif.
95 * This module was deprecated and has been removed. Users of chan_h323
96 should use chan_ooh323.
100 * This module was deprecated and has been removed. Users of chan_jingle
101 should use chan_motif.
105 * The SIPPEER dialplan function no longer supports using a colon as a
106 delimiter for parameters. The parameters for the function should be
107 delimited using a comma.
109 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
110 of the function should use the CHANNEL function instead.
114 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
115 Enabling PFS is attempted by default, and is dependent on the configuration
116 of the module using TLS.
117 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
118 specify a ECDHE cipher suite in sip.conf, for example:
119 tlscipher=AES128-SHA:DES-CBC3-SHA
120 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
121 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
122 default dh2048.pem - see
123 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
124 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
125 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
126 Consider re-ordering your cipher suites in the respective configuration
128 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
129 will use PFS when offered by the client. Clients which do not offer PFS
130 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
134 * The ast_channel_feature_hooks* functions have been added to allow features
135 such as DTMF hooks, interval hooks, and bridge event hooks to be made
136 available to a channel when the channel is bridged. Previously, these
137 features were provided exclusively by the caller of ast_bridge_join()
138 outside of "basic" type bridges.
142 * The JACK_HOOK function now supports audio with a sample rate higher than
147 * The SetMusicOnHold dialplan application was deprecated and has been removed.
148 Users of the application should use the CHANNEL function's musicclass
151 * The WaitMusicOnHold dialplan application was deprecated and has been
152 removed. Users of the application should use MusicOnHold with a duration
157 * The 'say' family of dialplan applications now support the Japanese
158 language. The 'language' parameter in say.conf now recognizes a setting of
159 'ja', which will enable Japanese language specific mechanisms for playing
160 back numbers, dates, and other items.
164 * VoiceMail and VoiceMailMain now support the Japanese language. The
165 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
166 which will enable prompts to be played back using a Japanese grammatical
167 structure. Additional prompts are necessary for this functionality,
169 - jb-arimasu: there is
170 - jb-arimasen: there is not
171 - jb-oshitekudasai: please press
179 * Added the ability to support PostgreSQL application_name on connections.
180 This allows PostgreSQL to display the configured name in the
181 pg_stat_activity view and CSV log entries. This setting is configurable
182 for res_config_pgsql via the dbappname configuration setting in
187 * New options to play a beep when starting a recording and stopping a recording
188 have been added. The option "p" will play a beep to the channel that starts
189 the recording. The option "P" will play a beep to the channel that stops the
192 ------------------------------------------------------------------------------
193 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
194 ------------------------------------------------------------------------------
198 * Stored recordings now support a new operation, copy. This will take an
199 existing stored recording and copy it to a new location in the recordings
204 * The endpoint configuration object now supports 'accountcode'. Any channel
205 created for an endpoint with this setting will have its accountcode set
206 to the specified value.
210 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
211 unconditionally inhereted through masquerades. As a side benefit, more
212 than one audiohook of a given type may persist through a masquerade now.
214 ------------------------------------------------------------------------------
215 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
216 ------------------------------------------------------------------------------
220 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
221 connect with an incoming caller after being alerted to the presence
222 of the incoming caller. The most likely reason this would happen is
223 the agent did not acknowledge the call in time.
227 * New events have been added for the TALK_DETECT function. When the function
228 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
229 emitted to connected AMI clients indicating the start/stop of talking on
234 * New event models have been aded for the TALK_DETECT function. When the
235 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
236 events will be emitted to connected WebSockets subscribed to the channel,
237 indicating the start/stop of talking on the channel.
241 * A new function, TALK_DETECT, has been added. When set on a channel, this
242 fucntion causes events indicating the starting/stoping of talking on said
243 channel to be emitted to both AMI and ARI clients.
245 ------------------------------------------------------------------------------
246 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
247 ------------------------------------------------------------------------------
251 * A new Playback URI 'tone' has been added. Tones are specified either as
252 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
253 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
254 URIs in that they must be stopped manually and will continue to occupy
255 a channel's ARI control queue until they are stopped. They also can not
256 be rewound or fastforwarded.
258 * User events can now be generated from ARI. Events can be signalled with
259 arbitrary json variables, and include one or more of channel, bridge, or
260 endpoint snapshots. An application must be specified which will receive
261 the event message (other applications can subscribe to it). The message
262 will also be delivered via AMI provided a channel is attached. Dialplan
263 generated user event messages are still transmitted via the channel, and
264 will only be received by a stasis application they are attached to or if
265 the channel is subscribed to.
269 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
270 fields for prohibited callingpres information. Values are legacy, no, and
271 yes. By default, legacy is used.
272 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
273 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
274 headers are appended to outbound SIP messages just as they are with
275 allowed callingpres values, but data about the remote party's identity is
277 When sendrpid=rpid, only the remote party's domain is anonymized.
278 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
279 headers are not sent.
280 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
281 party information in tact even for prohibited callingpres information.
282 In the case of PAI, a Privacy: id header will be appended for prohibited
283 calling information to communicate that the private information should
284 not be relayed to untrusted parties.
288 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
289 which can be used to announce the parked call's location to an arbitrary
290 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
291 parties in a one to one bridge, 'TimeoutChannel' is treated as having
292 parked 'Channel' like with the Park Call DTMF feature and will receive
293 announcements prior to being hung up.
295 ------------------------------------------------------------------------------
296 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
297 ------------------------------------------------------------------------------
300 --------------------------
301 * Record application now has an option 'o' which allows 0 to act as an exit
302 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
303 * Monitor() - A new option, B(), has been added that will turn on a periodic
304 beep while the call is being recorded.
307 --------------------------
308 * A new function was added: PERIODIC_HOOK. This allows running a periodic
309 dialplan hook on a channel. Any audio generated by this hook will be
310 injected into the call.
313 --------------------------
314 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
315 as the chanprefix parameter if the 'u' option is specified.
318 --------------------------
319 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
320 conference user menus.
322 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
323 menus, bridge settings, and user settings that have been applied by the
324 CONFBRIDGE dialplan function.
326 * The ConfBridge dialplan application now sets a channel variable,
327 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
328 how a channel exited the conference.
330 * Added conference user option 'announce_join_leave_review'. This option
331 implies 'announce_join_leave' with the added effect that the user will
332 be asked if they want to confirm or re-record the recording of their
333 name when entering the conference
336 --------------------------
337 * At exit, the Directory application now sets a channel variable
338 DIRECTORY_RESULT to one of the following based on the reason for exiting:
339 OPERATOR user requested operator by pressing '0' for operator
340 ASSISTANT user requested assistant by pressing '*' for assistant
341 TIMEOUT user pressed nothing and Directory stopped waiting
342 HANGUP user's channel hung up
343 SELECTED user selected a user from the directory and is routed
344 USEREXIT user pressed '#' from the selection prompt to exit
345 FAILED directory failed in a way that wasn't accounted for. Dang.
348 --------------------------
349 * MusicOnHold streams (all modes other than "files") now support wide band
353 --------------------------
354 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
355 and for the channel executing Page respectively.
358 --------------------------
359 * PickupChan now accepts channel uniqueids of channels to pickup.
362 --------------------------
363 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
364 to 'true' (case insensitive), then any Say application (SayNumber,
365 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
366 anticipate DTMF. If DTMF is received, these applications will behave like
367 the background application and jump to the received extension once a match
368 is established or after a short period of inactivity.
371 -------------------------
372 * A new function, MIXMONITOR, has been added to allow access to individual
373 instances of MixMonitor on a channel.
374 * A new option, B(), has been added that will turn on a periodic beep while the
375 call is being recorded.
378 -------------------------
381 -------------------------
382 * TEL URI support for inbound INVITE requests has been added. chan_sip will
383 now handle TEL schemes in the Request and From URIs. The phone-context in
384 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
388 -------------------------
389 * Core Show Locks output now includes Thread/LWP ID if the platform
390 supports this feature.
391 * New "logger add channel" and "logger remove channel" CLI commands have
392 been added to allow creation and deletion of dynamic logger channels
393 without configuration changes. These dynamic logger channels will only
394 exist until the next restart of asterisk.
398 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
399 the new AST_SORCERY diaplan function.
403 * The live recording object on recording events now contains a target_uri
404 field which contains the URI of what is being recorded.
406 * The bridge type used when creating a bridge is now a comma separated list of
407 bridge properties. Valid options are: mixing, holding, dtmf_events, and
410 * A channelId can now be provided when creating a channel, either in the
411 uri (POST channels/my-channel-id) or as query parameter. A local channel
412 will suffix the second channel id with ';2' unless provided as query
413 parameter otherChannelId.
415 * A bridgeId can now be provided when creating a bridge, either in the uri
416 (POST bridges/my-bridge-id) or as a query parameter.
418 * A playbackId can be provided when starting a playback, either in the uri
419 (POST channels/my-channel-id/play/my-playback-id /
420 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
422 * A snoop channel can be started with a snoopId, in the uri or query.
426 * Originate now takes optional parameters ChannelId and OtherChannelId,
427 used to set the UniqueId on creation. The other id is assigned to the
428 second channel when dialing LOCAL, or defaults to appending ;2 if only
429 the single Id is given.
431 * The Mixmonitor action now has a "Command" header that can be used to
432 indicate a post-process command to run once recording finishes.
436 * A new set of Alembic scripts has been added for CDR tables. This will create
437 a 'cdr' table with the default schema that Asterisk expects.
441 * A new module, res_hep, has been added, that acts as a generic packet
442 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
443 It can be configured via hep.conf. Other modules can use res_hep to send
444 message traffic to a HEP capture server.
448 * A new module, res_hep_pjsip, has been added that will forward PJSIP
449 message traffic to a HEP capture server. See res_hep for more
454 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
455 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
457 * Added the following new CLI commands:
458 - "pjsip show contacts" - list all current PJSIP contacts.
459 - "pjsip show contact" - show specific information about a current PJSIP
461 - "pjsip show channel" - show detailed information about a PJSIP channel.
465 * A new module, res_pjsip_multihomed handles situations where the system
466 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
467 determines which interface should be used during message sending.
469 res_pjsip_pidf_digium_body_supplement
471 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
472 request body formatting for presence support in Digium phones.
474 res_pjsip_send_to_voicemail
476 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
477 particular headers to transfer a PJSIP channel directly to a particular
478 extension that has VoiceMail. This is intended to be used with Digium
479 phones that support this feature.
481 res_pjsip_outbound_registration
483 * A new CLI command has been added: "pjsip show registrations", which lists
484 all configured PJSIP registrations
487 ------------------------------------------------------------------------------
488 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
489 ------------------------------------------------------------------------------
493 * Added a new module that provides AMI control over MWI within Asterisk,
494 res_mwi_external_ami. Note that this module depends on res_mwi_external;
495 for more information on enabling this module, see res_mwi_external.
496 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
497 the MWIGet/MWIGetComplete events.
499 * The DialStatus field in the DialEnd event can now contain additional
500 statuses that convey how the dial operation terminated. This includes
501 ABORT, CONTINUE, and GOTO.
503 * AMI will now emit security events. A new class authorization has been
504 added in manager.conf for the security events, 'security'. The new events
506 - FailedACL - raised when a request violates an ACL check
507 - InvalidAccountID - raised when a request fails an authentication
508 check due to an invalid account ID
509 - SessionLimit - raised when a request fails due to exceeding the
510 number of allowed concurrent sessions for a service
511 - MemoryLimit - raised when a request fails due to an internal memory
513 - LoadAverageLimit - raised when a request fails because a configured
514 load average limit has been reached
515 - RequestNotAllowed - raised when a request is not allowed by
517 - AuthMethodNotAllowed - raised when a request used an authentication
518 method not allowed by the service
519 - RequestBadFormat - raised when a request is received with bad formatting
520 - SuccessfulAuth - raised when a request successfully authenticates
521 - UnexpectedAddress - raised when a request has a different source address
522 then what is expected for a session already in progress with a service
523 - ChallengeResponseFailed - raised when a request's attempt to authenticate
524 has been challenged, and the request failed the authentication challenge
525 - InvalidPassword - raised when a request provides an invalid password
526 during an authentication attempt
527 - ChallengeSent - raised when an Asterisk service send an authentication
528 challenge to a request
529 - InvalidTransport - raised when a request attempts to use a transport not
530 allowed by the Asterisk service
532 * Bridge related events now have two additional fields: BridgeName and
533 BridgeCreator. BridgeName is a descriptive name for the bridge;
534 BridgeCreator is the name of the entity that created the bridge. This
535 affects the following events: ConfbridgeStart, ConfbridgeEnd,
536 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
537 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
538 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
542 * The Bridge data model now contains the additional fields 'name' and
543 'creator'. The 'name' field conveys a descriptive name for the bridge;
544 the 'creator' field conveys the name of the entity that created the bridge.
545 This affects all responses to HTTP requests that return a Bridge data model
546 as well as all event derived data models that contain a Bridge data model.
547 The POST /bridges operation may now optionally specify a name to give to
548 the bridge being created.
550 * Added a new ARI resource 'mailboxes' which allows the creation and
551 modification of mailboxes managed by external MWI. Modules res_mwi_external
552 and res_stasis_mailbox must be enabled to use this resource. For more
553 information on external MWI control, see res_mwi_external.
555 * Added new events for externally initiated transfers. The event
556 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
557 of a bridge in the ARI controlled application to the dialplan; the
558 BridgeAttendedTransfer event is raised when a channel initiates an
559 attended transfer of a bridge in the ARI controlled application to the
562 * Channel variables may now be specified as a body parameter to the
563 POST /channels operation. The 'variables' key in the JSON is interpreted
564 as a sequence of key/value pairs that will be added to the created channel
565 as channel variables. Other parameters in the JSON body are treated as
566 query parameters of the same name.
570 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
571 automatically handled by the HTTP server if a request is received with a
572 Transfer-Encoding type of "chunked".
576 * Path support has been added with the 'support_path' option in registration
579 * A 'debug' option has been added to the globals section that will allow
580 sip messages to be logged.
582 * A 'set_var' option has been added to endpoints that will automatically
583 set the desired variable(s) on a channel created for that endpoint.
585 * Several new tables and columns have been added to the realtime schema for
586 the res_pjsip related modules. See the UPGRADE.txt notes for updating
591 * A new module, res_mwi_external, has been added to Asterisk. This module
592 acts as a base framework that other modules can build on top of to allow
593 an external system to control MWI within Asterisk. For implementations
594 that make use of res_mwi_external, see res_mwi_external_ami and
595 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
596 that may produce MWI themselves, such as app_voicemail. res_mwi_external
597 and other modules that depend on it cannot be built or loaded with
598 app_voicemail present.
602 * DNS functionality will now automatically be enabled if the system configured
603 nameservers can be retrieved. If the system configured nameservers can not be
604 retrieved the functionality will resort to using system resolution. Functionalty
605 such as SRV records and failover will not be available if system resolution
608 ------------------------------------------------------------------------------
609 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
610 ------------------------------------------------------------------------------
615 Asterisk 12 is a standard release of the Asterisk project. As such, the
616 focus of development for this release was on core architectural changes and
617 major new features. This includes:
618 * A more flexible bridging core based on the Bridging API
619 * A new internal message bus, Stasis
620 * Major standardization and consistency improvements to AMI
621 * Addition of the Asterisk RESTful Interface (ARI)
622 * A new SIP channel driver, chan_pjsip
623 In addition, as the vast majority of bridging in Asterisk was migrated to the
624 Bridging API used by ConfBridge, major changes were made to most of the
625 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
627 Specifications have been written for the affected interfaces. These
628 specifications are available on the Asterisk wiki:
629 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
630 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
631 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
633 It is *highly* recommended that anyone migrating to Asterisk 12 read the
634 information regarding its release both in this file and in the accompanying
635 UPGRADE.txt file. More detailed information on the major changes can be found
636 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
641 * Added build option DISABLE_INLINE. This option can be used to work around a
642 bug in gcc. For more information, see
643 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
645 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
646 the CHANNEL_TRACE build option were incompatible with the new bridging
649 * Asterisk now optionally uses libxslt to improve XML documentation generation
650 and maintainability. If libxslt is not available on the system, some XML
651 documentation will be incomplete.
653 * Asterisk now depends on libjansson. If a package of libjansson is not
654 available on your distro, please see http://www.digip.org/jansson/.
656 * Asterisk now depends on libuuid and, optionally, uriparser. It is
657 recommended that you install uriparser, even if it is optional.
659 * The new SIP stack and channel driver uses a particular version of PJSIP.
660 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
661 configuring and installing PJSIP for usage with Asterisk.
663 * Optional API was re-implemented to be more portable, and no longer requires
664 weak reference support from the compiler. The build option OPTIONAL_API may
665 be disabled to disable Optional API support.
672 * Along with AgentRequest, this application has been modified to be a
673 replacement for chan_agent. The act of a channel calling the AgentLogin
674 application places the channel into a pool of agents that can be
675 requested by the AgentRequest application. Note that this application, as
676 well as all other agent related functionality, is now provided by the
677 app_agent_pool module. See chan_agent and AgentRequest for more information.
679 * This application no longer performs agent authentication. If authentication
680 is desired, the dialplan needs to perform this function using the
681 Authenticate or VMAuthenticate application or through an AGI script before
684 * If this application is called and the agent is already logged in, the
685 dialplan will continue exection with the AGENT_STATUS channel variable set
686 to ALREADY_LOGGED_IN.
688 * The agents.conf schema has changed. Rather than specifying agents on a
689 single line in comma delineated fashion, each agent is defined in a separate
690 context. This allows agents to use the power of context templates in their
693 * A number of parameters from agents.conf have been removed. This includes
694 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
695 urlprefix, and savecallsin. These options were obsoleted by the move from
696 a channel driver model to the bridging/application model provided by
701 * A new application, this will request a logged in agent from the pool and
702 bridge the requested channel with the channel calling this application.
703 Logged in agents are those channels that called the AgentLogin application.
704 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
705 application will be set with an appropriate error value.
709 * This application has been removed. It was a holdover from when
710 AgentCallbackLogin was removed.
714 * Added support for additional Ademco DTMF signalling formats, including
715 Express 4+1, Express 4+2, High Speed and Super Fast.
717 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
718 call time, in milliseconds, to run the application.
720 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
721 maximum number of times to retry the call.
723 * Added a new configuration option answait. If set, the AlarmReceiver
724 application will wait the number of milliseconds specified by answait
725 after the channel has answered. Valid values range between 500
726 milliseconds and 10000 milliseconds.
728 * Added configuration option no_group_meta. If enabled, grouping of metadata
729 information in the AlarmReceiver log file will be skipped.
733 * It is now no longer possible to bypass updating the CDR on the channel
734 when answering. CDRs reflect the state of the channel and will always
735 reflect the time they were Answered.
739 * A new application in Asterisk, this will place the calling channel
740 into a holding bridge, optionally entertaining them with some form of
741 media. Channels participating in a holding bridge do not interact with
742 other channels in the same holding bridge. Optionally, however, a channel
743 may join as an announcer. Any media passed from an announcer channel is
744 played to all channels in the holding bridge. Channels leave a holding
745 bridge either when an optional timer expires, or via the ChannelRedirect
746 application or AMI Redirect action.
750 * All participants in a bridge can now be kicked out of a conference room
751 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
752 command, i.e., 'confbridge kick <conference> all'
754 * CLI output for the 'confbridge list' command has been improved. When
755 displaying information about a particular bridge, flags will now be shown
756 for the participating users indicating properties of that user.
758 * The ConfbridgeList event now contains the following fields: WaitMarked,
759 EndMarked, and Waiting. This displays additional properties about the
760 user's profile, as well as whether or not the user is waiting for a
761 Marked user to enter the conference.
763 * Added a new option for conference recording, record_file_append. If enabled,
764 when the recording is stopped and then re-started, the existing recording
765 will be used and appended to.
767 * ConfBridge now has the ability to set the language of announcements to the
768 conference. The language can be set on a bridge profile in confbridge.conf
769 or by the dialplan function CONFBRIDGE(bridge,language)=en.
773 * The channel variable CPLAYBACKSTATUS may now return the value
774 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
775 such as AMI. See the AMI action ControlPlayback for more information.
779 * Added the 'a' option, which allows the caller to enter in an additional
780 alias for the user in the directory. This option must be used in conjunction
781 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
782 specified in voicemail.conf.
786 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
787 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
788 containing the unique ID of the bridge that the channel happens to be in.
792 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
793 for more information.
795 * Variables are no longer purged from the original CDR. See the 'v' option for
798 * The 'A' option has been removed. The Answer time on a CDR is never updated
801 * The 'd' option has been removed. The disposition on a CDR is a function of
802 the state of the channel and cannot be altered.
804 * The 'D' option has been removed. Who the Party B is on a CDR is a function
805 of the state of the respective channels involved in the CDR and cannot be
808 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
809 such that the start time and, if applicable, the answer time was updated.
810 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
811 'r' option now triggers the Reset, setting the start time (and answer time
812 if applicable) to the current time. Note that the 'a' option still sets
813 the answer time to the current time if the channel was already answered.
815 * The 's' option has been removed. A variable can be set on the original CDR
816 if desired using the CDR function, and removed from a forked CDR using the
819 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
820 longer applies in the CDR engine.
822 * The 'v' option now prevents the copy of the variables from the original CDR
823 to the forked CDR. Previously the variables were always copied but were
824 removed from the original. This was changed as removing variables from a CDR
825 can have unintended side effects - this option allows the user to prevent
826 propagation of variables from the original to the forked without modifying
831 * Added the 'n' option to MeetMe to prevent application of the DENOISE
832 function to a channel joining a conference. Some channel drivers that vary
833 the number of audio samples in a voice frame will experience significant
834 quality problems if a denoiser is attached to the channel; this option gives
835 them the ability to remove the denoiser without having to unload func_speex.
839 * The 'b' option now includes conferences as well as sounds played to the
842 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
843 running during a transfer. If a MixMonitor is started on a channel,
844 the MixMonitor will continue to record the audio passing through the
845 channel even in the presence of transfers.
849 * The NoCDR application is deprecated. Please use the CDR_PROP function to
852 * While the NoCDR application will prevent CDRs for a channel from being
853 propagated to registered CDR backends, it will not prevent that data from
854 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
855 function that enables CDRs on a channel will restore those records that have
856 not yet been finalized.
860 * The app_parkandannounce module has been removed. The application
861 ParkAndAnnounce is now provided by the res_parking module. See the
862 res_parking changes for more information.
866 * Added queue available hint. The hint can be added to the dialplan using the
867 following syntax: exten,hint,Queue:{queue_name}_avail
868 For example, if the name of the queue is 'markq':
869 exten => 8501,hint,Queue:markq_avail
870 This will report 'InUse' if there are no logged in agents or no free agents.
871 It will report 'Idle' when an agent is free.
873 * Queues now support a hint for member paused state. The hint uses the form
874 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
875 are the name of the queue and the name of the member to subscribe to,
876 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
877 Members will show as In Use when paused.
879 * The configuration options eventwhencalled and eventmemberstatus have been
880 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
881 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
882 sent. The "Variable" fields will also no longer exist on the Agent* events.
883 These events can be filtered out from a connected AMI client using the
884 eventfilter setting in manager.conf.
886 * The queue log now differentiates between blind and attended transfers. A
887 blind transfer will result in a BLINDTRANSFER message with the destination
888 context and extension. An attended transfer will result in an
889 ATTENDEDTRANSFER message. This message will indicate the method by which
890 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
891 for running an application on a bridge or channel, or "LINK" for linking
892 two bridges together with local channels. The queue log will also now detect
893 externally initiated blind and attended transfers and record the transfer
896 * When performing queue pause/unpause on an interface without specifying an
897 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
898 least one member of any queue exists for that interface.
900 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
901 for realtime queue log entries.
905 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
906 CDRs when they were previously disabled on a channel.
908 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
909 backends occurs on an as-needed basis in order to preserve linkedid
910 propagation and other needed behavior.
914 * A new application, this is similar to SayAlpha except that it supports
915 case sensitive playback of the specified characters. For example,
916 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
920 * This application is deprecated in favor of CHANNEL(amaflags).
924 * The SendDTMF application will now accept 'W' as valid input. This will cause
925 the application to delay one second while streaming DTMF.
929 * A new application in Asterisk 12, this hands control of the channel calling
930 the application over to an external system. Currently, external systems
931 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
935 * UserEvent will now handle duplicate keys by overwriting the previous value
938 * In addition to AMI, UserEvent invocations will now be distributed to any
939 interested Stasis applications.
943 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
944 system as mailbox@context. The rest of the system cannot add @default
945 to mailbox identifiers for app_voicemail that do not specify a context
946 any longer. It is a mailbox identifier format that should only be
947 interpreted by app_voicemail.
949 * The voicemail.conf configuration file now has an 'alias' configuration
950 parameter for use with the Directory application. The voicemail realtime
951 database table schema has also been updated with an 'alias' column.
956 * Pass through support has been added for both VP8 and Opus.
958 * Added format attribute negotiation for the Opus codec. Format attribute
959 negotiation is provided by the res_format_attr_opus module.
964 * Masquerades as an operation inside Asterisk have been effectively hidden
965 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
966 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
967 dropping of frame/audio hooks, and other internal implementation details
968 that users had to deal with. This fundamental change has large implications
969 throughout the changes documented for this version. For more information
970 about the new core architecture of Asterisk, please see the Asterisk wiki.
972 * Multiple parties in a bridge may now be transferred. If a participant in a
973 multi-party bridge initiates a blind transfer, a Local channel will be used
974 to execute the dialplan location that the transferer sent the parties to. If
975 a participant in a multi-party bridge initiates an attended transfer,
976 several options are possible. If the attended transfer results in a transfer
977 to an application, a Local channel is used. If the attended transfer results
978 in a transfer to another channel, the resulting channels will be merged into
981 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
982 driver specific. If the channel variable is set on the transferrer channel,
983 the sound will be played to the target of an attended transfer.
985 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
986 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
987 listed. Any more peers in the bridge will not be included in the list.
988 BRIDGEPEER is not valid in holding bridges like parking since those channels
989 do not talk to each other even though they are in a bridge.
991 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
992 and will contain a value if the BRIDGEPEER's channel driver supports it.
994 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
995 was responsible for an attended transfer in a similar fashion to
998 * Modules using the Configuration Framework or Sorcery must have XML
999 configuration documentation. This configuration documentation is included
1000 with the rest of Asterisk's XML documentation, and is accessible via CLI
1001 commands. See the CLI changes for more information.
1003 AMI (Asterisk Manager Interface)
1005 * Major changes were made to both the syntax as well as the semantics of the
1006 AMI protocol. In particular, AMI events have been substantially improved
1007 in this version of Asterisk. For more information, please see the AMI
1008 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1010 * AMI events that reference a particular channel or bridge will now always
1011 contain a standard set of fields. When multiple channels or bridges are
1012 referenced in an event, fields for at least some subset of the channels
1013 and bridges in the event will be prefixed with a descriptive name to avoid
1014 name collisions. See the AMI event documentation on the Asterisk wiki for
1017 * The CLI command 'manager show commands' no longer truncates command names
1018 longer than 15 characters and no longer shows authorization requirement
1019 for commands. 'manager show command' now displays the privileges needed
1020 for using a given manager command instead.
1022 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1023 peer in its response if the peer has a subscribe context set.
1025 * The SIPqualifypeer action now acknowledges the request once it has
1026 established that the request is against a known peer. It also issues a new
1027 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1029 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1030 specifies the duration of the digit to be played, in milliseconds.
1032 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1033 updates when changes occur instead of requiring the use of pollmailboxes.
1035 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1036 AMI client to manipulate audio currently being played back on a channel. The
1037 supported operations depend on the application being used to send audio to
1038 the channel. When the audio playback was initiated using the ControlPlayback
1039 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1040 stopped, restarted, reversed, or skipped forward. When initiated by other
1041 mechanisms (such as the Playback application), the audio can be stopped,
1042 reversed, or skipped forward.
1044 * Channel related events now contain a snapshot of channel state, adding new
1045 fields to many of these events.
1047 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1048 in a future release. Please use the common 'Exten' field instead.
1050 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1051 fields. The channel state fields will come before the body fields.
1053 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1054 'UnParkedCall' have changed significantly in the new res_parking module.
1056 The 'Channel' and 'From' headers are gone. For the channel that was parked
1057 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1058 has a number of fields associated with it. The old 'Channel' header relayed
1059 the same data as the new 'ParkeeChannel' header.
1061 The 'From' field was ambiguous and changed meaning depending on the event.
1062 for most of these, it was the name of the channel that parked the call
1063 (the 'Parker'). There is no longer a header that provides this channel name,
1064 however the 'ParkerDialString' will contain a dialstring to redial the
1065 device that parked the call.
1067 On UnParkedCall events, the 'From' header would instead represent the
1068 channel responsible for retrieving the parkee. It receives a channel
1069 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1072 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1074 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1075 fashion has changed the field names 'StartExten' and 'StopExten' to
1076 'StartSpace' and 'StopSpace' respectively.
1078 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1079 manager.conf has been removed.
1081 * Channel Variables conveyed with a channel no longer contain the name of the
1082 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1083 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1084 event, the various ChanVariable fields will contain a suffix that specifies
1085 which channel they correspond to.
1087 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1088 event always conveys the AMI event for a particular channel.
1090 * All 'Reload' events have been consolidated into a single event type. This
1091 event will always contain a Module field specifying the name of the module
1092 and a Status field denoting the result of the reload. All modules now issue
1093 this event when being reloaded.
1095 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1096 fail to receive this event due to being connected after modules have loaded.
1097 AMI connections that want to know when Asterisk is ready should listen for
1098 the 'FullyBooted' event.
1100 * app_fax now sends the same send fax/receive fax events as res_fax. The
1101 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1102 now the 'ReceiveFAX' event.
1104 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1105 'MusicOnHoldStop'. The sub type field has been removed.
1107 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1108 carrier for another protocol.
1110 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1111 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1112 to the specific channel. 'Both' may be specified to play a tone to both
1113 channels. The old 'yes' option is still accepted as a way of playing the
1114 tone to Channel2 only.
1116 * The AMI 'Status' response event to the AMI Status action replaces the
1117 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1118 indicate what bridge the channel is currently in.
1120 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1121 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1124 * The AMI events in app_queue have been made more consistent with each other.
1125 Events that reference channels (QueueCaller* and Agent*) will show
1126 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1127 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1129 * The 'MCID' AMI event now publishes a channel snapshot when available and
1130 its non-channel-snapshot parameters now use either the "MCallerID" or
1131 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1132 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1133 parameters in the channel snapshot.
1135 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1136 'AgentLogin' and 'AgentLogoff' respectively.
1138 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1139 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1141 * 'ChannelUpdate' events have been removed.
1143 * All AMI events now contain a 'SystemName' field, if available.
1145 * Local channel optimization is now conveyed in two events:
1146 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1147 when the Local channel driver begins attempting to optimize itself out of
1148 the media path; the End event is sent after the channel halves have
1149 successfully optimized themselves out of the media path.
1151 * Local channel information in events is now prefixed with 'LocalOne' and
1152 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1153 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1154 and 'LocalOptimizationEnd' events.
1156 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1157 account. When set in the general context, it will act as the default
1158 setting for defined accounts.
1160 * The 'BridgeAction' event was removed. It technically added no value, as the
1161 Bridge Action already receives confirmation of the bridge through a
1162 successful completion Event.
1164 * The 'BridgeExec' events were removed. These events duplicated the events that
1165 occur in the Briding API, and are conveyed now through BridgeCreate,
1166 BridgeEnter, and BridgeLeave events.
1168 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1169 previous versions. They now report all SR/RR packets sent/received, and
1170 have been restructured to better reflect the data sent in a SR/RR. In
1171 particular, the event structure now supports multiple report blocks.
1173 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1174 raised when a blind transfer/attended transfer completes successfully.
1175 They contain information about the transfer that just completed, including
1176 the location of the transfered channel.
1178 * Added a 'security' class to AMI which outputs the required fields for
1179 security messages similar to the log messages from res_security_log
1181 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1182 that describes the status value in a human readable string.
1184 CDR (Call Detail Records)
1186 * Significant changes have been made to the behavior of CDRs. The CDR engine
1187 was effectively rewritten and built on the Stasis message bus. For a full
1188 definition of CDR behavior in Asterisk 12, please read the specification
1189 on the Asterisk wiki (wiki.asterisk.org).
1191 * CDRs will now be created between all participants in a bridge. For each
1192 pair of channels in a bridge, a CDR is created to represent the path of
1193 communication between those two endpoints. This lets an end user choose who
1194 to bill for what during bridge operations with multiple parties.
1196 * The duration, billsec, start, answer, and end times now reflect the times
1197 associated with the current CDR for the channel, as opposed to a cumulative
1198 measurement of all CDRs for that channel.
1200 * When a CDR is dispatched, user defined CDR variables from both parties are
1201 included in the resulting CDR. If both parties have the same variable, only
1202 the Party A value is provided.
1204 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1205 information regarding the CDR engine is logged as verbose messages. This
1206 option should only be used if the behavior of the CDR engine needs to be
1209 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1210 normally configured in cdr.conf.
1212 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1213 specified, this command provides a summary of the channels with CDR
1214 information and their statistics. When {channel} is specified, it shows
1215 detailed information about all records associated with {channel}.
1217 CEL (Channel Event Logging)
1219 * CEL has undergone significant rework in Asterisk 12, and is now built on the
1220 Stasis message bus. Please see the specification for CEL on the Asterisk
1221 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
1224 * The 'extra' field of all CEL events that use it now consists of a JSON blob
1225 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
1227 * BLINDTRANSFER events now report the transferee bridge unique
1228 identifier, extension, and context in a JSON blob as the extra string
1229 instead of the transferee channel name as the peer.
1231 * ATTENDEDTRANSFER events now report the peer as NULL and additional
1232 information in the 'extra' string as a JSON blob. For transfers that occur
1233 between two bridged channels, the 'extra' JSON blob contains the primary
1234 bridge unique identifier, the secondary channel name, and the secondary
1235 bridge unique identifier. For transfers that occur between a bridged channel
1236 and a channel running an app, the 'extra' JSON blob contains the primary
1237 bridge unique identifier, the secondary channel name, and the app name.
1239 * LOCAL_OPTIMIZE events have been added to convey local channel
1240 optimizations with the record occurring for the semi-one channel and
1241 the semi-two channel name in the peer field.
1243 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
1244 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
1245 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
1246 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
1247 regardless of whether or not that bridge happens to contain multiple
1252 * When compiled with '--enable-dev-mode', the astobj2 library will now add
1253 several CLI commands that allow for inspection of ao2 containers that
1254 register themselves with astobj2. The CLI commands are 'astobj2 container
1255 dump', 'astobj2 container stats', and 'astobj2 container check'.
1257 * Added specific CLI commands for bridge inspection. This includes 'bridge
1258 show all', which lists all bridges in the system, and 'bridge show {id}',
1259 which provides specific information about a bridge.
1261 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
1262 ejecting the channels currently in the bridge. If the channels cannot
1263 continue in the dialplan or application that put them in the bridge, they
1266 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1268 * Added commands to inspect and manipulate the registered bridge technologies.
1269 This include 'bridge technology show', which lists the registered bridge
1270 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1271 which controls whether or not a registered bridge technology can be used
1272 during smart bridge operations. If a technology is suspended, it will not
1273 be used when a bridge technology is picked for channels; when unsuspended,
1274 it can be used again.
1276 * The command 'config show help {module} {type} {option}' will show
1277 configuration documentation for modules with XML configuration
1278 documentation. When {module}, {type}, and {option} are omitted, a listing
1279 of all modules with registered documentation is displayed. When {module}
1280 is specified, a listing of all configuration types for that module is
1281 displayed, along with their synopsis. When {module} and {type} are
1282 specified, a listing of all configuration options for that type are
1283 displayed along with their synopsis. When {module}, {type}, and {option}
1284 are specified, detailed information for that configuration option is
1287 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1288 a listing of all installed media sounds available on the system and
1289 detailed information about a sound, respectively.
1291 * 'xmldoc dump' has been added. This CLI command will dump the XML
1292 documentation DOM as a string to the specified file. The Asterisk core
1293 will populate certain XML elements pulled from the source files with
1294 additional run-time information; this command lets a user produce the
1295 XML documentation with all information.
1299 * Parking has been pulled from core and placed into a separate module called
1300 res_parking. See Parking changes below for more details. Configuration for
1301 parking should now be performed in res_parking.conf. Configuration for
1302 parking in features.conf is now unsupported.
1304 * Core attended transfers now have several new options. While performing an
1305 attended transfer, the transferer now has the following options:
1306 - *1 - cancel the attended transfer (configurable via atxferabort)
1307 - *2 - complete the attended transfer, dropping out of the call
1308 (configurable via atxfercomplete)
1309 - *3 - complete the attended transfer, but stay in the call. This will turn
1310 the call into a multi-party bridge (configurable via atxferthreeway)
1311 - *4 - swap to the other party. Once an attended transfer has begun, this
1312 options may be used multiple times (configurable via atxferswap)
1314 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1315 must be on the channel initiating the transfer to have any effect.
1317 * The BRIDGE_FEATURES channel variable would previously only set features for
1318 the calling party and would set this feature regardless of whether the
1319 feature was in caps or in lowercase. Use of a caps feature for a letter
1320 will now apply the feature to the calling party while use of a lowercase
1321 letter will apply that feature to the called party.
1323 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1325 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1326 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1327 activated the dynamic feature.
1329 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1330 only on the channel executing the dynamic feature. Executing a dynamic
1331 feature on the bridge peer in a multi-party bridge will execute it on all
1332 peers of the activating channel.
1334 * You can now have the settings for a channel updated using the FEATURE()
1335 and FEATUREMAP() functions inherited to child channels by setting
1336 FEATURE(inherit)=yes.
1338 * automixmon now supports additional channel variables from automon including:
1339 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1340 and TOUCH_MIXMONITOR_MESSAGE_STOP
1342 * A new general features.conf option 'recordingfailsound' has been added which
1343 allowssetting a failure sound for a user tries to invoke a recording feature
1344 such as automon or automixmon and it fails.
1346 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1347 features.c for atxferdropcall=no to work properly. This option now just
1352 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1353 be used. Given that this can cause the Asterisk log files to grow quickly,
1354 this option should only be used if an external mechanism for log management
1359 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1360 will store the path information for that peer when it registers. Realtime
1361 tables can also use the 'supportpath' field to enable Path header support.
1363 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1364 objectIdentifier. This maps to the supportpath option in sip.conf.
1368 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1369 provides modules a useful abstraction on top of the many storage mechanisms
1370 in Asterisk, including the Asterisk Database, static configuration files,
1371 static Realtime, and dynamic Realtime. It also provides a caching service.
1372 Users can configure a hierarchy of data storage layers for specific modules
1375 * All future modules which utilize Sorcery for object persistence must have a
1376 column named "id" within their schema when using the Sorcery realtime module.
1377 This column must be able to contain a string of up to 128 characters in length.
1379 Security Events Framework
1381 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1382 the "seconds-microseconds" format that it was using previously.
1386 * The Stasis message bus is a publish/subscribe message bus internal to
1387 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1388 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1389 Stasis can be configured in stasis.conf. Note that these parameters operate
1390 at a very low level in Asterisk, and generally will not require changes.
1394 * When a channel driver is configured to enable jiterbuffers, they are now
1395 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1396 is already set for that channel when it enters, such as by the JITTERBUFFER
1397 function, then the existing jitterbuffer will be used and the one set by
1398 the channel driver will not be applied.
1402 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1403 dialplan applications provided by the app_agent_pool module. Agents are
1404 connected with callers using the new AgentRequest dialplan application.
1405 The Agents:<agent-id> device state is available to monitor the status of an
1406 agent. See agents.conf.sample for valid configuration options.
1408 * The updatecdr option has been removed. Altering the names of channels on a
1409 CDR is not supported - the name of the channel is the name of the channel,
1410 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1411 has also been removed, for the same reason.
1413 * The endcall and enddtmf configuration options are removed. Use the
1414 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1415 channel before calling AgentLogin.
1419 * chan_bridge has been removed. Its functionality has been incorporated
1420 directly into the ConfBridge application itself.
1424 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1425 of the specified span and its B-channels. Note that this command should
1426 only be used if you understand the risks it entails.
1428 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1429 A range of channels can be specified to be destroyed. Note that this command
1430 should only be used if you understand the risks it entails.
1432 * Added the CLI command 'dahdi create channels'. A range of channels can be
1433 specified to be created, or the keyword 'new' can be used to add channels
1436 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1437 the exact configured mailbox name. For app_voicemail mailboxes this is
1440 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1444 * IPv6 support has been added. We are now able to bind to and
1445 communicate using IPv6 addresses.
1449 * The /b option has been removed.
1451 * chan_local moved into the system core and is no longer a loadable module.
1455 * Added general support for busy detection.
1457 * Added ECAM command support for Sony Ericsson phones.
1461 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1462 SIP stack. A collection of resource modules provides the bulk of the SIP
1463 functionality. For more information on the new SIP channel driver, see
1464 https://wiki.asterisk.org/wiki/x/JYGLAQ
1468 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1469 using the 'supportpath' setting, either on a global basis or on a peer basis.
1470 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1471 set of proxies by using a pre-loaded route-set defined by the Path headers in
1472 the REGISTER request. See Realtime updates for more configuration information.
1474 * The SIP_CODEC family of variables may now specify more than one codec. Each
1475 codec must be separated by a comma. The first codec specified is the
1476 preferred codec for the offer. This allows a dialplan writer to specify both
1477 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1479 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1480 in the core, and can be filtered out using the 'eventfilter' parameter
1483 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1484 codecs configured for a peer instead of the requested codec.
1486 * The option "register_retry_403" has been added to chan_sip to work around
1487 servers that are known to erroneously send 403 in response to valid
1488 REGISTER requests and allows Asterisk to continue attepmting to connect.
1492 * Added the 'immeddialkey' parameter. If set, when the user presses the
1493 configured key the already entered number will be immediately dialed. This
1494 is useful when the dialplan allows for variable length pattern matching.
1495 Valid options are '*' and '#'.
1497 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1498 milliseconds) before a call forward is considered to not be answered.
1500 * The 'serviceurl' parameter allows Service URLs to be attached to line
1509 * The password option has been disabled, as the AgentLogin application no
1510 longer provides authentication.
1514 * Due to changes in the Asterisk core, this function is no longer needed to
1515 preserve a MixMonitor on a channel during transfer operations and dialplan
1516 execution. It is effectively obsolete.
1520 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1521 deprecated. Use the CHANNEL function instead to access these attributes.
1523 * The 'l' option has been removed. When reading a CDR attribute, the most
1524 recent record is always used. When writing a CDR attribute, all non-finalized
1527 * The 'r' option has been removed, for the same reason as the 'l' option.
1529 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1534 * A new function CDR_PROP has been added. This function lets you set properties
1535 on a channel's active CDRs. This function is write-only. Properties accept
1536 boolean values to set/clear them on the channel's CDRs. Valid properties
1538 - 'party_a' - make this channel the preferred Party A in any CDR between two
1539 channels. If two channels have this property set, the creation time of the
1540 channel is used to determine who is Party A. Note that dialed channels are
1541 never Party A in a CDR.
1542 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1543 application when set to True, and analogous to the 'e' option in ResetCDR
1548 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1549 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1550 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1553 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1554 string, i.e., [[context],extension],priority. If set on a channel, if a
1555 channel leaves a bridge but is not hung up it will resume dialplan execution
1560 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1561 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1562 The value of this setting is ignored when disabled is used for the argument.
1566 * A new function provided by chan_pjsip, this function can be used in
1567 conjunction with the Dial application to construct a dial string that will
1568 dial all contacts on an Address of Record associated with a chan_pjsip
1573 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1574 outbound channel prior to dialing.
1578 * Redirecting reasons can now be set to arbitrary strings. This means
1579 that the REDIRECTING dialplan function can be used to set the redirecting
1580 reason to any string. It also allows for custom strings to be read as the
1581 redirecting reason from SIP Diversion headers.
1585 * The SPEECH_ENGINE function now supports read operations. When read from, it
1586 will return the current value of the requested attribute.
1590 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1591 system as mailbox@context. The rest of the system cannot add @default
1592 to mailbox identifiers for app_voicemail that do not specify a context
1593 any longer. It is a mailbox identifier format that should only be
1594 interpreted by app_voicemail.
1600 res_agi (Asterisk Gateway Interface)
1602 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1604 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1607 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1608 will start the playback of the audio at the position specified. It will
1609 also return the final position of the file in 'endpos'.
1611 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1612 channel variable if the user stopped the file playback or if a remote
1613 entity stopped the playback. If neither stopped the playback, it will
1614 indicate the overall success/failure of the playback. If stopped early,
1615 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1618 * The SAY ALPHA command now accepts an additional parameter to control
1619 whether it specifies the case of uppercase, lowercase, or all letters to
1620 provide functionality similar to SayAlphaCase.
1622 res_ari (Asterisk RESTful Interface) (and others)
1624 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1625 control telephony primitives in Asterisk by remote client. This includes
1626 channels, bridges, endpoints, media, and other fundamental concepts. Users
1627 of ARI can develop their own communications applications, controlling
1628 multiple channels using an HTTP RESTful interface and receiving JSON events
1629 about the objects via a WebSocket connection. ARI can be configured in
1630 Asterisk via ari.conf. For more information on ARI, see
1631 https://wiki.asterisk.org/wiki/x/0YCLAQ
1635 * Parking has been extracted from the Asterisk core as a loadable module,
1636 res_parking. Configuration for parking is now provided by res_parking.conf.
1637 Configuration through features.conf is no longer supported.
1639 * res_parking uses the configuration framework. If an invalid configuration is
1640 supplied, res_parking will fail to load or fail to reload. Previously,
1641 invalid configurations would generally be accepted, with certain errors
1642 resulting in individually disabled parking lots.
1644 * Parked calls are now placed in bridges. While this is largely an
1645 architectural change, it does have implications on how channels in a parking
1646 lot are viewed. For example, commands that display channels in bridges will
1647 now also display the channels in a parking lot.
1649 * The order of arguments for the new parking applications have been modified.
1650 Timeout and return context/exten/priority are now implemented as options,
1651 while the name of the parking lot is now the first parameter. See the
1652 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1653 in-depth information as well as syntax.
1655 * Extensions are by default no longer automatically created in the dialplan to
1656 park calls or pickup parked calls. Generation of dialplan extensions can be
1657 enabled using the 'parkext' configuration option.
1659 * ADSI functionality for parking is no longer supported. The 'adsipark'
1660 configuration option has been removed as a result.
1662 * The PARKINGSLOT channel variable has been deprecated in favor of
1663 PARKING_SPACE to match the naming scheme of the new system.
1665 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1666 channel even when the configuration option 'comebactoorigin' is enabled.
1668 * A new CLI command 'parking show' has been added. This allows a user to
1669 inspect the parking lots that are currently in use.
1670 'parking show <parkinglot>' will also show the parked calls in a specific
1673 * The CLI command 'parkedcalls' is now deprecated in favor of
1674 'parking show <parkinglot>'.
1676 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1677 can be used to get a list of parked calls for a specific parking lot.
1679 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1680 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1681 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1682 longer a required argument.
1684 * The ParkAndAnnounce application is now provided through res_parking instead
1685 of through the separate app_parkandannounce module.
1687 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1688 by default. Instead, it will follow the timeout rules of the parking lot. The
1689 old behavior can be reproduced by using the 'c' option.
1691 * Dynamic parking lots will now fail to be created under the following
1693 - if the parking lot specified by PARKINGDYNAMIC does not exist
1694 - if they require exclusive park and parkedcall extensions which overlap
1695 with existing parking lots.
1697 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1698 currently contain no calls. Dynamic parking lots containing parked calls
1699 will persist through the reloads without alteration.
1701 * If 'parkext_exclusive' is set for a parking lot and that extension is
1702 already in use when that parking lot tries to register it, this is now
1703 considered a parking system configuration error. Configurations which do
1704 this will be rejected.
1706 * Added channel variable PARKER_FLAT. This contains the name of the extension
1707 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1708 comebacktoorigin is disabled, but the dialplan or an external control
1709 mechanism wants to use the extension in the park-dial context that was
1710 generated to re-dial the parker on timeout.
1712 res_pjsip (and many others)
1714 * A large number of resource modules make up the SIP stack based on pjsip.
1715 The chan_pjsip channel driver users these resource modules to provide
1716 various SIP functionality in Asterisk. The majority of configuration for
1717 these modules is performed in pjsip.conf. Other modules may use their
1718 own configuration files.
1720 * Added 'set_var' option for an endpoint. For each variable specified that
1721 variable gets set upon creation of a channel involving the endpoint.
1725 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1726 them, an Asterisk-specific version of PJSIP needs to be installed.
1727 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1729 res_statsd/res_chan_stats
1731 * A new resource module, res_statsd, has been added, which acts as a statsd
1732 client. This module allows Asterisk to publish statistics to a statsd
1733 server. In conjunction with res_chan_stats, it will publish statistics about
1734 channels to the statsd server. It can be configured via res_statsd.conf.
1738 * Device state for XMPP buddies is now available using the following format:
1739 XMPP/<client name>/<buddy address>
1740 If any resource is available the device state is considered to be not in use.
1741 If no resources exist or all are unavailable the device state is considered
1748 Realtime/Database Scripts
1750 * Asterisk previously included example db schemas in the contrib/realtime/
1751 directory of the source tree. This has been replaced by a set of database
1752 migrations using the Alembic framework. This allows you to use alembic to
1753 initialize the database for you. It will also serve as a database migration
1754 tool when upgrading Asterisk in the future.
1756 See contrib/ast-db-manage/README.md for more details.
1760 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1761 This python script will convert an existing sip.conf file to a
1762 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1763 is meant to be an aid in converting an existing chan_sip configuration to
1764 a chan_pjsip configuration, but it is expected that configuration beyond
1765 what the script provides will be needed.
1767 ------------------------------------------------------------------------------
1768 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1769 ------------------------------------------------------------------------------
1773 * The Asterisk build system will now build and install a shared library
1774 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1775 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1776 that Asterisk can ensure that these functions do *not* get called by any
1777 modules that are loaded into Asterisk, since they should only be called once
1778 in any single process. If desired, this feature can be disabled by supplying
1779 the "--disable-asteriskssl" option to the configure script.
1781 * A new make target, 'full', has been added to the Makefile. This performs
1782 the same compilation actions as make all, but will also scan the entirety of
1783 each source file for documentation. This option is needed to generate AMI
1784 event documentation. Note that your system must have Python in order for
1785 this make target to succeed.
1787 * The optimization portion of the build system has been reworked to avoid
1788 broken builds on certain architectures. All architecture-specific
1789 optimization has been removed in favor of using -march=native to allow gcc
1790 to detect the environment in which it is running when possible. This can
1791 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1793 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1794 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1796 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1797 previously parsed the header file to obtain the version of Asterisk, you
1798 will now have to go through Asterisk to get the version information.
1806 * Added 'F()' option. Similar to the dial option, this can be supplied with
1807 arguments indicating where the callee should go after the caller is hung up,
1808 or without options specified, the priority after the Queue will be used.
1813 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1814 all non-admin participants on a conference. The confbridge configuration
1815 file also allows for the default sounds played to all conference users when
1816 this occurs to be overriden using sound_participants_unmuted and
1817 sound_participants_muted.
1819 * Added menu action participant_count. This will playback the number of
1820 current participants in a conference.
1822 * Added announcement configuration option to user profile. If set the sound
1823 file will be played to the user, and only the user, upon joining the
1826 * Added record_file_append option that defaults to "yes", but if set to no
1827 will create a new file between each start/stop recording.
1832 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1833 channels respectively before the callee channels are called.
1838 * Added support for IPv6.
1840 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1841 external process will cause the current playlist to be cleared, including
1842 stopping any audio file that is currently playing. This is useful when you
1843 want to interrupt audio playback only when specific DTMF is entered by the
1849 * A new option, 'I' has been added to app_followme. By setting this option,
1850 Asterisk will not update the caller with connected line changes when they
1851 occur. This is similar to app_dial and app_queue.
1853 * The 'N' option is now ignored if the call is already answered.
1855 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1856 and caller channels respectively before the callee channels are called.
1858 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1864 * MixMonitor hooks now have IDs associated with them which can be used to
1865 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1866 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1867 now accepts that ID as an argument.
1869 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1870 indicated mailboxes.
1875 * The connect action in app_mysql now allows you to specify a port number to
1876 connect to. This is useful if you run a MySQL server on a non-standard
1882 * Increased the default number of allowed destinations from 5 to 12.
1887 * The app_page application now no longer depends on DAHDI or app_meetme. It
1888 has been re-architected to use app_confbridge internally.
1893 * Added queue options autopausebusy and autopauseunavail for automatically
1894 pausing a queue member when their device reports busy or congestion.
1896 * The 'ignorebusy' option for queue members has been deprecated in favor of
1897 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1898 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1899 per interface basis. Individual ringinuse values can now be set in
1900 queues.conf via an argument to member definitions. Lastly, the queue
1901 'ringinuse' setting now only determines defaults for the per member
1902 'ringinuse' setting and does not override per member settings like it does
1903 in earlier versions.
1905 * Added 'F()' option. Similar to the dial option, this can be supplied with
1906 arguments indicating where the callee should go after the caller is hung up,
1907 or without options specified, the priority after the Queue will be used.
1909 * Added new option log_member_name_as_agent, which will cause the membername to
1910 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1911 state_interface has been set.
1913 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1915 * App_queue will now play periodic announcements for the caller that
1916 holds the first position in the queue while waiting for answer.
1920 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1921 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1922 changed arguments to SayUnixTime so that every option is truly optional even
1923 when using multiple options (so that j option could be used without having to
1924 manually specify timezone and format) There are other benefits, e.g., format
1925 can now be used without specifying time zone as well.
1930 * Addition of the VM_INFO function - see Function changes.
1932 * The imapserver, imapport, and imapflags configuration options can now be
1933 overriden on a user by user basis.
1935 * When voicemail plays a message's envelope with saycid set to yes, when
1936 reaching the caller id field it will play a recording of a file with the same
1937 base name as the sender's callerid if there is a similarly named file in
1938 <astspooldir>/recordings/callerids/
1940 * Voicemails now contains a unique message identifier "msg_id", which is stored
1941 in the message envelope with the sound files. IMAP backends will now store
1942 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1943 backends will store the message identifier in a "msg_id" column. See
1944 UPGRADE.txt for more information.
1946 * Added VoiceMailPlayMsg application. This application will play a single
1947 voicemail message from a mailbox. The result of the application, SUCCESS or
1948 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1953 * Hangup handlers can be attached to channels using the CHANNEL() function.
1954 Hangup handlers will run when the channel is hung up similar to the h
1955 extension. The hangup_handler_push option will push a GoSub compatible
1956 location in the dialplan onto the channel's hangup handler stack. The
1957 hangup_handler_pop option will remove the last added location, and optionally
1958 replace it with a new GoSub compatible location. The hangup_handler_wipe
1959 option will remove all locations on the stack, and optionally add a new
1962 * The expression parser now recognizes the ABS() absolute value function,
1963 which will convert negative floating point values to positive values.
1965 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1966 control of faxdetect.
1968 * Addition of the VM_INFO function that can be used to retrieve voicemail
1969 user information, such as the email address and full name.
1970 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1973 * The REDIRECTING function now supports the redirecting original party id
1976 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1977 lets you set some of the configuration options from the [general] section
1978 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1979 the key sequence used to activate built-in features, such as blindxfer,
1980 and automon. See the built-in documentation for details.
1982 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1983 instead of simply the uri. This is the format that MessageSend() can use
1984 in the from parameter for outgoing SIP messages.
1986 * Added the PRESENCE_STATE function. This allows retrieving presence state
1987 information from any presence state provider. It also allows setting
1988 presence state information from a CustomPresence presence state provider.
1989 See AMI/CLI changes for related commands.
1991 * Added the AMI_CLIENT function to make manager account attributes available
1992 to the dialplan. It currently supports returning the current number of
1993 active sessions for a given account.
1995 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1996 and the REDIRECTING functions.
2004 * Added a manager event "LocalBridge" for local channel call bridges between
2005 the two pseudo-channels created.
2010 * Added dialtone_detect option for analog ports to disconnect incoming
2011 calls when dialtone is detected.
2013 * Added option colp_send to send ISDN connected line information. Allowed
2014 settings are block, to not send any connected line information; connect, to
2015 send connected line information on initial connect; and update, to send
2016 information on any update during a call. Default is update.
2018 * Add options namedcallgroup and namedpickupgroup to support installations
2019 where a higher number of groups (>64) is required.
2021 * Added support to use private party ID information with PRI calls.
2026 * A new channel driver named chan_motif has been added which provides support for
2027 Google Talk and Jingle in a single channel driver. This new channel driver includes
2028 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2029 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2030 specification, current Google Jingle specification, and the original Google Talk
2036 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2037 globally and overriden on a peer by peer basis.
2039 * Direct media functionality has been added. Options in config are:
2040 directmedia (directrtp) and directrtpsetup (earlydirect)
2042 * ChannelUpdate events now contain a CallRef header.
2047 * Asterisk will no longer substitute CID number for CID name in the display
2048 name field if CID number exists without a CID name. This change improves
2049 compatibility with certain device features such as Avaya IP500's directory
2052 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2053 created using that setting to not be removed during SIP reload.
2055 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2056 request with a "Record:" header, this will turn the requested feature on/off.
2057 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2058 dynamic features must be enabled and configured properly on the requesting
2059 channel for this to function properly.
2061 * Add support to realtime for the 'callbackextension' option.
2063 * When multiple peers exist with the same address, but differing
2064 callbackextension options, incoming requests that are matched by address
2065 will be matched to the peer with the matching callbackextension if it is
2068 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2069 which set the force_rport and comedia options automatically if Asterisk
2070 detects that an incoming SIP request crossed a NAT after being sent by
2071 the remote endpoint.
2073 * The default global nat setting in sip.conf has been changed from force_rport
2074 to auto_force_rport.
2076 * NAT settings are now a combinable list of options. The equivalent of the
2077 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2079 * Adds an option send_diversion which can be disabled to prevent
2080 diversion headers from automatically being added to INVITE requests.
2082 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2083 be sent to the remote host at a given interval to keep the NAT mapping open.
2084 This can be enabled using the keepalive configuration option.
2086 * Add option 'tonezone' to specify country code for indications. This option
2087 can be set both globally and overridden for specific peers.
2089 * The SIP Security Events Framework now supports IPv6.
2091 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2092 between multiple user agents. When set, for directmedia reinvites,
2093 Asterisk will not send an immediate reinvite on an incoming call leg. This
2094 option is useful when peered with another SIP user agent that is known to
2095 send immediate direct media reinvites upon call establishment.
2097 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2100 * Add options subminexpiry and submaxexpiry to set limits of subscription
2101 timer independently from registration timer settings. The setting of the
2102 registration timer limits still is done by options minexpiry, maxexpiry
2103 and defaultexpiry. For backwards compatibility the setting of minexpiry
2104 and maxexpiry also is used to configure the subscription timer limits if
2105 subminexpiry and submaxexpiry are not set in sip.conf.
2107 * Set registration timer limits to default values when reloading sip
2108 configuration and values are not set by configuration.
2110 * Add options namedcallgroup and namedpickupgroup to support installations
2111 where a higher number of groups (>64) is required.
2113 * When a MESSAGE request is received, the address the request was received from
2114 is now saved in the SIP_RECVADDR variable.
2116 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2117 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2118 the ANI2/OLI information is set on the channel, which can be retrieved using
2119 the CALLERID function.
2121 * Peers can now be configured to support negotiation of ICE candidates using
2122 the setting icesupport. See res_rtp_asterisk changes for more information.
2124 * Added support for format attribute negotiation. See the Codecs changes for
2127 * Extra headers specified with SIPAddHeader are sent with the REFER message
2128 when using Transfer application. See refer_addheaders in sip.conf.sample.
2130 * Added support to use private party ID information with calls.
2132 * Adds an option discard_remote_hold_retrieval that when set stops telling
2133 the peer to start music on hold.
2138 * Added skinny version 17 protocol support.
2142 --------------------
2143 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2145 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2146 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2147 as per the UNISTIM protocol.
2149 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2150 as well as tx. Also fixed issue with call "Timer" displaying as French "Durée"
2152 * Added ability to use multiple lines for a single phone. This allows multiple
2153 calls to occur on a single phone, using callwaiting and switching between calls.
2155 * Added option 'sharpdial' allowing end dialing by pressing # key
2157 * Added option 'interdigit_timer' to control phone dial timeout
2159 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2161 * Added global 'debug' option, that enables debug in channel driver
2163 * Added ability to translate on-screen menu in multiple languages. Tested on
2164 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2165 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2168 * In addition to English added French and Russian languages for on-screen menus
2170 * Reworked dialing number input: added dialing by timeout, immediate dial on
2171 on dialplan compare, phone number length now not limited by screen size
2173 * Added ability to pickup a call using features.conf defined value and
2179 * Add options namedcallgroup and namedpickupgroup to support installations
2180 where a higher number of groups (>64) is required.
2182 * Added support to use private party ID information with calls.
2187 * The minimum DTMF duration can now be configured in asterisk.conf
2188 as "mindtmfduration". The default value is (as before) set to 80 ms.
2189 (previously it was only available in source code)
2191 * Named ACLs can now be specified in acl.conf and used in configurations that
2192 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2193 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2194 working ACL. In addition, some CLI commands have been added to provide
2195 show information and allow for module reloading - see CLI Changes.
2197 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2198 items (separated by commas), and items in the rule can be negated by prefixing
2199 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2200 longer necessray to control the order that the 'permit' and 'deny' columns are
2201 returned from queries.
2203 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2204 be used within the dynamic weight attribute when specifying a mapping.
2206 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2207 header, instead of putting the user defined event name there. When enabled
2208 the UserDefType header is added for user defined events. This feature is
2209 enabled with the setting show_user_defined.
2211 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2212 line purposes use the following variables instead of their macro equivalents:
2213 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2214 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
2215 cc_callback_macro in channel configurations.
2217 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
2220 * Call files now support the "early_media" option to connect with an outgoing
2221 extension when early media is received.
2223 * Added support to use private party ID information with calls.
2228 * A new channel variable, AGIEXITONHANGUP, has been added which allows
2229 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
2230 AGI application would exit immediately after a channel hangup is detected.
2232 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
2233 are resolved and each address is attempted in turn until one succeeds or
2237 AMI (Asterisk Manager Interface)
2239 * The originate action now has an option "EarlyMedia" that enables the
2240 call to bridge when we get early media in the call. Previously,
2241 early media was disregarded always when originating calls using AMI.
2243 * Added setvar= option to manager accounts (much like sip.conf)
2245 * Originate now generates an error response if the extension given is not found
2248 * MixMonitor will now show IDs associated with the mixmonitor upon creating
2249 them if the i(variable) option is used. StopMixMonitor will accept
2250 MixMonitorID as an option to close specific MixMonitors.
2252 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
2253 updated to include information about peers configured with
2254 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
2255 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
2256 returned if auto_force_rport is not enabled.
2258 * Added SIPpeerstatus manager command which will generate PeerStatus events
2259 similar to the existing PeerStatus events found in chan_sip on demand.
2261 * Hangup now can take a regular expression as the Channel option. If you want
2262 to hangup multiple channels, use /regex/ as the Channel option. Existing
2263 behavior to hanging up a single channel is unchanged, but if you pass a regex,
2264 the manager will send you a list of channels back that were hung up.
2266 * Support for IPv6 addresses has been added.
2268 * AMI Events can now be documented in the Asterisk source. Note that AMI event
2269 documentation is only generated when Asterisk is compiled using 'make full'.
2270 See the CLI section for commands to display AMI event information.
2272 * The AMI Hangup event now includes the AccountCode header so you can easily
2273 correlate with AMI Newchannel events.
2275 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2276 the StateInterface of the queue member.
2278 * Added AMI event SessionTimeout in the Call category that is issued when a
2279 call is terminated due to either RTP stream inactivity or SIP session timer
2282 * CEL events can now contain a user defined header UserDefType. See core
2283 changes for more information.
2285 * OOH323 ChannelUpdate events now contain a CallRef header.
2287 * Added PresenceState command. This command will report the presence state for
2288 the given presence provider.
2290 * Added Parkinglots command. This will list all parking lots as a series of
2291 AMI Parkinglot events.
2293 * Added MessageSend command. This behaves in the same manner as the
2294 MessageSend application, and is a technolgoy agnostic mechanism to send out
2295 of call text messages.
2297 * Added "message" class authorization. This grants an account permission to
2298 send out of call messages. Write-only.
2303 * The "dialplan add include" command has been modified to create context a context
2304 if one does not already exist. For instance, "dialplan add include foo into bar"
2305 will create context "bar" if it does not already exist.
2307 * A "dialplan remove context" command has been added to remove a context from
2310 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2311 filenames of all running mixmonitors on a channel.
2313 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2314 numeric instead of 0, 1, or 2.
2316 * "stun show status" will show a table describing how the STUN client is
2319 * "acl show [named acl]" will show information regarding a Named ACL. The
2320 acl module can be reloaded with "reload acl".
2322 * Added CLI command to display AMI event information - "manager show events",
2323 which shows a list of all known and documented AMI events, and "manager show
2324 event [event name]", which shows detail information about a specific AMI
2327 * The result of the CLI command "queue show" now includes the state interface
2328 information of the queue member.
2330 * The command "core set verbose" will now set a separate level of logging for
2331 each remote console without affecting any other console.
2333 * Added command "cdr show pgsql status" to check connection status
2335 * "sip show channel" will now display the complete route set.
2337 * Added "presencestate list" command. This command will list all custom
2338 presence states that have been set by using the PRESENCE_STATE dialplan
2341 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2342 command. This changes a custom presence to a new state.
2347 * Codec lists may now be modified by the '!' character, to allow succinct
2348 specification of a list of codecs allowed and disallowed, without the
2349 requirement to use two different keywords. For example, to specify all
2350 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2352 * Add support for parsing SDP attributes, generating SDP attributes, and
2353 passing it through. This support includes codecs such as H.263, H.264, SILK,
2354 and CELT. You are able to set up a call and have attribute information pass.
2355 This should help considerably with video calls.
2357 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2358 just like the GSM codec.
2362 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2363 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2367 * Asterisk version and build information is now logged at the beginning of a
2370 * Threads belonging to a particular call are now linked with callids which get
2371 added to any log messages produced by those threads. Log messages can now be
2372 easily identified as involved with a certain call by looking at their call id.
2373 Call ids may also be attached to log messages for just about any case where
2374 it can be determined to be related to a particular call.
2376 * Each logging destination and console now have an independent notion of the
2377 current verbosity level. Logger.conf now allows an optional argument to
2378 the 'verbose' specifier, indicating the level of verbosity sent to that
2379 particular logging destination. Additionally, remote consoles now each
2380 have their own verbosity level. The command 'core set verbose' will now set
2381 a separate level for each remote console without affecting any other
2387 * Added 'announcement' option which will play at the start of MOH and between
2388 songs in modes of MOH that can detect transitions between songs (eg.
2394 * New per parking lot options: comebackcontext and comebackdialtime. See
2395 configs/features.conf.sample for more details.
2397 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2400 * Channel variable PARKEDCALL is now set with the name of the parking lot
2401 when a timeout occurs.
2407 CDR Postgresql Driver
2409 * Added command "cdr show pgsql status" to check connection status
2412 CDR Adaptive ODBC Driver
2414 * Added schema option for databases that support specifying a schema.
2422 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2423 CALENDAR_WRITE has completed successfully.
2428 * A new option, 'probation' has been added to rtp.conf
2429 RTP in strictrtp mode can now require more than 1 packet to exit learning
2430 mode with a new source (and by default requires 4). The probation option
2431 allows the user to change the required number of packets in sequence to any
2432 desired value. Use a value of 1 to essentially restore the old behavior.
2433 Also, with strictrtp on, Asterisk will now drop all packets until learning
2434 mode has successfully exited. These changes are based on how pjmedia handles
2435 media sources and source changes.
2437 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2438 enabled or disabled using the icesupport setting. A variety of other
2439 settings have been introduced to configure STUN/TURN connections.
2444 * A new module, res_corosync, has been introduced. This module uses the
2445 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2446 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2447 Device State (presence) information. This module is very similar to, and
2448 is a replacement for the res_ais module that was in previous releases of
2454 * This module adds a cleaned up, drop-in replacement for res_jabber called
2455 res_xmpp. This provides the same externally facing functionality but is
2456 implemented differently internally. res_jabber has been deprecated in favor
2457 of res_xmpp; please see the UPGRADE.txt file for more information.
2462 * The safe_asterisk script has been updated to allow several of its parameters
2463 to be set from environment variables. This also enables a custom run
2464 directory of Asterisk to be specified, instead of defaulting to /tmp.
2466 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2467 its value to determine the directory to assume is the top-level directory of
2468 the source tree. If the variable is not set, it defaults to the current
2469 behavior and uses the current working directory.
2471 ------------------------------------------------------------------------------
2472 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2473 ------------------------------------------------------------------------------
2477 * Asterisk now has protocol independent support for processing text messages
2478 outside of a call. Messages are routed through the Asterisk dialplan.
2479 SIP MESSAGE and XMPP are currently supported. There are options in
2480 jabber.conf and sip.conf to allow enabling these features.
2481 -> jabber.conf: see the "sendtodialplan" and "context" options.
2482 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2483 and "outofcall_message_context" options.
2484 The MESSAGE() dialplan function and MessageSend() application have been
2485 added to go along with this functionality. More detailed usage information
2486 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2487 * If real-time text support (T.140) is negotiated, it will be preferred for
2488 sending text via the SendText application. For example, via SIP, messages
2489 that were once sent via the SIP MESSAGE request would be sent via RTP if
2490 T.140 text is negotiated for a call.
2494 * parkedmusicclass can now be set for non-default parking lots.
2496 Asterisk Manager Interface
2497 --------------------------
2498 * PeerStatus now includes Address and Port.
2499 * Added Hold events for when the remote party puts the call on and off hold
2500 for chan_dahdi ISDN channels.
2501 * Added new action MeetmeListRooms to list active conferences (shows same
2502 data as "meetme list" at the CLI).
2503 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2504 Description field that is set by 'description' in the channel configuration
2506 * Added Uniqueid header to UserEvent.
2507 * Added new action FilterAdd to control event filters for the current session.
2508 This requires the system permission and uses the same filter syntax as
2509 filters that can be defined in manager.conf
2510 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2511 versions had some instances of the event converted, but others were left
2512 as-is. All Unlink events should now be converted to Bridge events. The AMI
2513 protocol version number was incremented to 1.2 as a result of this change.
2515 Asterisk HTTP Server
2516 --------------------------
2517 * The HTTP Server can bind to IPv6 addresses.
2520 --------------------------
2521 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2522 with busydetect. usage example: busypattern=200,200,200,600
2525 --------------------------
2526 * New 'gtalk show settings' command showing the current settings loaded from
2528 * The 'logger reload' command now supports an optional argument, specifying an
2529 alternate configuration file to use.
2530 * 'dialplan add extension' command will now automatically create a context if
2531 the specified context does not exist with a message indicated it did so.
2532 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2533 Description field which can be populated with 'description' in the channel
2534 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2537 --------------------------
2538 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2539 thus allowing records which do NOT match the specified filter.
2540 * Added ability to log CONGESTION calls to CDR
2543 --------------------------
2544 * Ability to define custom SILK formats in codecs.conf.
2545 * Addition of speex32 audio format with translation.
2546 * CELT codec pass-through support and ability to define
2547 custom CELT formats in codecs.conf.
2548 * Ability to read raw signed linear files with sample rates
2549 ranging from 8khz - 192khz. The new file extensions introduced
2550 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2551 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2552 Skinny, H.323, etc) can still only support the following codecs:
2553 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2554 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2555 Video: h261, h263, h263p, h264, mpeg4
2560 --------------------------
2561 * New highly optimized and customizable ConfBridge application capable of
2562 mixing audio at sample rates ranging from 8khz-96khz.
2563 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2564 and bridge profiles on a channel.
2565 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2566 about a conference such as locked status and number of parties, admins,
2568 * Addition of video_mode option in confbridge.conf for adding video support
2569 into a bridge profile.
2570 * Addition of the follow_talker video_mode in confbridge.conf. This video
2571 mode dynamically switches the video feed to always display the loudest talker
2572 supplying video in the conference.
2576 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2577 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2578 variables from asterisk.conf.
2582 * Addition of the JITTERBUFFER dialplan function. This function allows
2583 for jitterbuffering to occur on the read side of a channel. By using
2584 this function conference applications such as ConfBridge and MeetMe can
2585 have the rx streams jitterbuffered before conference mixing occurs.
2586 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2588 * Added STRREPLACE function. This function let's the user search a variable
2589 for a given string to replace with another string as many times as the
2590 user specifies or just throughout the whole string.
2591 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2592 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2593 * Added extensions to chan_ooh323 in function CHANNEL()
2595 libpri channel driver (chan_dahdi) DAHDI changes
2596 --------------------------
2597 * Added moh_signaling option to specify what to do when the channel's bridged
2598 peer puts the ISDN channel on hold.
2599 * Added display_send and display_receive options to control how the display ie
2600 is handled. To send display text from the dialplan use the SendText()
2601 application when the option is enabled.
2602 * Added mcid_send option to allow sending a MCID request on a span.
2605 --------------------------
2606 * Added setvar option to calendar.conf to allow setting channel variables on
2607 notification channels.
2608 * Added "calendar show types" CLI command to list registered calendar
2612 --------------------------
2613 * Added two new options, r and t with file name arguments to record
2614 single direction (unmixed) audio recording separate from the bidirectional
2615 (mixed) recording. The mixed file name argument is optional now as long
2616 as at least one recording option is used.
2619 --------------------------
2620 * Added a new option, l, which will disable local call optimization for
2621 channels involved with the FollowMe thread. Use this option to improve
2622 compatability for a FollowMe call with certain dialplan apps, options, and
2626 --------------------------
2627 * Added option "k" that will automatically close the conference when there's
2628 only one person left when a user exits the conference.
2631 --------------------------
2632 * cel_pgsql now supports the 'extra' column for data added using the
2633 CELGenUserEvent() application.
2636 --------------------------
2637 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2638 in the sample extensions.lua file for syntax details.
2639 * Applications that perform jumps in the dialplan such as Goto will now
2640 execute properly. When pbx_lua detects that the context, extension, or
2641 priority we are executing on has changed it will immediately return control
2642 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2643 the priority after the currently executing priority.
2644 * An autoservice is now started by default for pbx_lua channels. It can be
2645 stopped and restarted using the autoservice_stop() and autoservice_start()
2649 --------------------------
2650 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2651 into a FAXStatus event with an 'Operation' header that will be either
2652 'send', 'receive', and 'gateway'.
2653 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2654 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2655 feature will handle converting a fax call between an audio T.30 fax terminal
2656 and an IFP T.38 fax terminal.
2660 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2661 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2662 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2666 * Added general option negative_penalty_invalid default off. when set
2667 members are seen as invalid/logged out when there penalty is negative.
2668 for realtime members when set remove from queue will set penalty to -1.
2669 * Added queue option autopausedelay when autopause is enabled it will be
2670 delayed for this number of seconds since last successful call if there
2671 was no prior call the agent will be autopaused immediately.
2672 * Added member option ignorebusy this when set and ringinuse is not
2673 will allow per member control of multiple calls as ringinuse does for
2678 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2680 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2681 one participant left (much like a normal call bridge)
2682 * Added extra argument to Originate to set timeout.
2686 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2687 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2688 utility in the UTILS section of menuselect. If an existing astdb is found and no
2689 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2690 convert an existing astdb to the SQLite3 version automatically at runtime.
2694 * Modules marked as deprecated are no longer marked as building by default. Enabling
2695 these modules is still available via menuselect.
2699 * authdebug is now disabled by default. To enable this functionaility again
2700 set authdebug = yes in iax.conf.
2704 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2705 releases it was disabled.
2709 * The PBX core previously made a call with a non-existing extension test for
2710 extension s@default and jump there if the extension existed.
2711 This was a bad default behaviour and violated the principle of least surprise.
2712 It has therefore been changed in this release. It may affect some
2713 applications and configurations that rely on this behaviour. Most channel
2714 drivers have avoided this for many releases by testing whether the extension
2715 called exists before starting the PBX and generating a local error.
2716 This behaviour still exists and works as before.
2718 Extension "s" is used when no extension is given in a channel driver,
2719 like immediate answer in DAHDI or calling to a domain with no user part
2722 ------------------------------------------------------------------------------
2723 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2724 ------------------------------------------------------------------------------
2728 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2729 now defaults to force_rport. It is very important that phones requiring nat=no be
2730 specifically set as such instead of relying on the default setting. If at all
2731 possible, all devices should have nat settings configured in the general section as
2732 opposed to configuring nat per-device.
2733 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2734 codecs sent in response to an INVITE to the single most preferred codec.
2735 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2736 to be used for the outgoing call. It must be one of the codecs configured
2738 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2739 to be used for holding a private key. If tlsprivatekey is not specified,
2740 tlscertfile is searched for both public and private key.
2741 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2742 outbound client connections to be specified.
2743 * The sendrpid parameter has been expanded to include the options
2744 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2745 header to be sent (equivalent to setting sendrpid=yes) and setting
2746 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2747 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2748 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2749 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2750 will accept the SDP even if the SDP version number is not properly incremented,
2751 but will generate a warning in the log indicating that the SIP peer that sent
2752 the SDP should have the 'ignoresdpversion' option set.
2753 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2754 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2755 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2756 remote side requests it and disables symmetric RTP support. Setting it to
2757 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2758 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2759 and enables symmetric RTP support.
2760 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2761 response. This permits the master channel to know how each channel dialled
2762 in a multi-channel setup resolved in an individual way. This carries a
2763 performance penalty and can be disabled in sip.conf using the
2764 'storesipcause' option.
2765 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2766 configuration for the externip and externhost options when tcp or tls is used.
2767 * Added support for message body (stored in content variable) to SIP NOTIFY message
2768 accessible via AMI and CLI.
2769 * Added 'media_address' configuration option which can be used to explicitly specify
2770 the IP address to use in the SDP for media (audio, video, and text) streams.
2771 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2772 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2774 * Added 'use_q850_reason' configuration option for generating and parsing
2775 if available Reason: Q.850;cause=<cause code> header. It is implemented
2776 in some gateways for better passing PRI/SS7 cause codes via SIP.
2777 * When dialing SIP peers, a new component may be added to the end of the dialstring
2778 to indicate that a specific remote IP address or host should be used when dialing
2779 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2780 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2781 ability to selectively force bridged channels to also be encrypted is also
2782 implemented. Branching in the dialplan can be done based on whether or not
2783 a channel has secure media and/or signaling.
2784 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2786 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2787 Charge messages to snom phones.
2788 * Added support for G.719 media streams.
2789 * Added support for 16khz signed linear media streams.
2790 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2791 RTP has been outfitted with the same abilities.
2792 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2793 available in device configurations as well as in the dial plan.
2794 * Addition of the 'subscribe_network_change' option for turning on and off
2795 res_stun_monitor module support in chan_sip.
2796 * Addition of the 'auth_options_requests' option for turning on and off
2797 authentication for OPTIONS requests in chan_sip.
2801 * Add #tryinclude statement for config files. This provides the same
2802 functionality as the #include statement however an asterisk module will
2803 still load if the filename does not exist. Using the #include statement
2804 Asterisk will not allow the module to load.
2808 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2809 on realtime updates.
2810 * Added the ability for chan_iax2 to inform the dialplan whether or not
2811 encryption is being used. This interoperates with the SIP SRTP implementation
2812 so that a secure SIP call can be bridged to a secure IAX call when the
2813 dialplan requires bridged channels to be "secure".
2814 * Addition of the 'subscribe_network_change' option for turning on and off
2815 res_stun_monitor module support in chan_iax.
2820 * Added ability to preset channel variables on indicated lines with the setvar
2821 configuration option. Also, clearvars=all resets the list of variables back
2823 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2824 See configs/res_pktccops.conf for more information.
2826 XMPP Google Talk/Jingle changes
2827 -------------------------------
2828 * Added the externip option to gtalk.conf.
2829 * Added the stunaddr option to gtalk.conf which allows for the automatic
2830 retrieval of the external ip from a stun server.
2834 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2835 match to a partial channel name.
2836 * Added .m3u support for Mp3Player application.
2837 * Added progress option to the app_dial D() option. When progress DTMF is
2838 present, those values are sent immediately upon receiving a PROGRESS message
2839 regardless if the call has been answered or not.
2840 * Added functionality to the app_dial F() option to continue with execution
2841 at the current location when no parameters are provided.
2842 * Added the 'a' option to app_dial to answer the calling channel before any
2843 announcements or macros are executed.
2844 * Modified app_dial to set answertime when the called channel answers even if
2845 the called channel hangs up during playback of an announcement.
2846 * Modified app_dial 'r' option to support an additional parameter to play an
2847 indication tone from indications.conf
2848 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2849 to cycle through the next available channel. By default this is still '*'.
2850 * Added x() option to app_chanspy. This option allows DTMF to be set to
2851 exit the application.
2852 * The Voicemail application has been improved to automatically ignore messages
2853 that only contain silence.
2854 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2855 associated mailbox(es) to be greetings-only.
2856 * The ChanSpy application now has the 'S' option, which makes the application
2857 automatically exit once it hits a point where no more channels are available
2859 * The ChanSpy application also now has the 'E' option, which spies on a single
2860 channel and exits when that channel hangs up.
2861 * The MeetMe application now turns on the DENOISE() function by default, for
2862 each participant. In our tests, this has significantly decreased background
2863 noise (especially noisy data centers).
2864 * Voicemail now permits storage of secrets in a separate file, located in the
2865 spool directory of each individual user. The control for this is located in
2866 the "passwordlocation" option in voicemail.conf. Please see the sample
2867 configuration for more information.
2868 * The ChanIsAvail application now exposes the returned cause code using a separate
2869 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2870 * Added 'd' option to app_followme. This option disables the "Please hold"
2872 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2873 received will terminate recording.
2874 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2875 Previously the folder could only be set per context, but has now been extended
2876 using the imapfolder option.
2877 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2878 * Voicemail now allows the pager date format to be specified separately from the
2880 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2881 to allow joining, leaving, and sending text to group chats.
2882 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2883 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2884 to all paged phones (and optionally excluding the caller's one using the new
2885 option 'n') before the call is bridged.
2886 * The 'f' option to Dial has been augmented to take an optional argument. If no
2887 argument is provided, the 'f' option works as it always has. If an argument is
2888 provided, then the connected party information of all outgoing channels created
2889 during the Dial will be set to the argument passed to the 'f' option.
2890 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2892 * The OSP lookup application adds in/outbound network ID, optional security,
2893 number portability, QoS reporting, destination IP port, custom info and service
2895 * Added new application VMSayName that will play the recorded name of the voicemail
2896 user if it exists, otherwise will play the mailbox number.
2897 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2898 retrieve state for a particular bridge, where <name> is the conference name
2899 * app_directory now allows exiting at any time using the operator or pound key.
2900 * Voicemail now supports setting a locale per-mailbox.
2901 * Two new applications are provided for declining counting phrases in multiple
2902 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2904 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2906 * Voicemail now includes rdnis within msgXXXX.txt file.
2907 * ExternalIVR now supports IPv6 addresses.
2908 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2909 at https://wiki.asterisk.org/wiki/x/oQBB
2910 * ParkedCall and Park can now specify the parking lot to use.
2914 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2915 over SRV records associated with a specific service. From the CLI, type
2916 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2917 details on how these may be used.
2918 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2919 pitch of a channel's tx and rx audio streams.
2920 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2921 setting various connected line and redirecting party information.
2922 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2923 support ISDN subaddressing.
2924 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2925 * For DAHDI channels, the CHANNEL() dialplan function now allows
2926 the dialplan to request changes in the configuration of the active
2927 echo canceller on the channel (if any), for the current call only.
2930 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2932 The possible values are:
2934 on - normal mode (the echo canceller is actually reinitialized)
2936 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2938 voice - voice mode (returns from FAX mode, reverting the changes that
2939 were made when FAX mode was requested)
2940 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2941 and setting variables on the channel which created the current channel.
2942 Administrators should take care to avoid naming conflicts, when multiple
2943 channels are dialled at once, especially when used with the Local channel
2944 construct (which all could set variables on the master channel). Usage
2945 of the HASH() dialplan function, with the key set to the name of the slave
2946 channel, is one approach that will avoid conflicts.
2947 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2949 * func_odbc now allows multiple row results to be retrieved without using
2950 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2951 from the same query by using the name of the function which retrieved the
2952 first row as an argument to ODBC_FETCH().
2953 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2954 dialplan. This function returns the content of the received message.
2955 * Added REPLACE, which searches a given variable name for a set of characters,
2956 then either replaces them with a single character or deletes them.
2957 * Added PASSTHRU, which literally passes the same argument back as its return
2958 value. The intent is to be able to use a literal string argument to
2959 functions that currently require a variable name as an argument.
2960 * HASH-associated variables now can be inherited across channel creation, by
2961 prefixing the name of the hash at assignment with the appropriate number of
2962 underscores, just like variables.
2963 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2964 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2965 whether or not channels that are bridged to the current channel will be
2966 required to have secure signaling and/or media.
2967 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2968 the current channel has secure signaling and/or media.
2969 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2970 "no_media_path" option.
2971 Returns "0" if there is a B channel associated with the call.
2972 Returns "1" if no B channel is associated with the call. The call is either
2973 on hold or is a call waiting call.
2974 * Added option to dialplan function CDR(), the 'f' option
2975 allows for high resolution times for billsec and duration fields.
2976 * FILE() now supports line-mode and writing.
2977 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2978 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2982 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2983 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2984 and is set when a dynamic feature is triggered.
2985 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2986 to dynamically create a new parking lot matching the value this varible is
2988 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2989 features.conf that should be the base for dynamic parkinglots.
2990 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2991 parkinglot should have.
2992 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2993 parkinglot should have.
2994 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2999 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3000 timeout has expired.
3001 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3002 to the caller when an Agent's phone is ringing. This can be used to indicate
3003 to the caller that their call is about to be picked up, which is nice when
3004 one has been on hold for an extened period of time.
3005 * A new config option, penaltymemberslimit, has been added to queues.conf.
3006 When set this option will disregard penalty settings when a queue has too
3008 * A new option, 'I' has been added to both app_queue and app_dial.
3009 By setting this option, Asterisk will not update the caller with
3010 connected line changes or redirecting party changes when they occur.
3011 * A 'relative-periodic-announce' option has been added to queues.conf. When
3012 enabled, this option will cause periodic announce times to be calculated
3013 from the end of announcements rather than from the beginning.
3014 * The autopause option in queues.conf can be passed a new value, "all." The
3015 result is that if a member becomes auto-paused, he will be paused in all
3016 queues for which he is a member, not just the queue that failed to reach
3018 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3019 * The queue logger now allows events to optionally propagate to a file,
3020 even when realtime logging is turned on. Additionally, realtime logging
3021 supports sending the event arguments to 5 individual fields, although it
3022 will fallback to the previous data definition, if the new table layout is
3025 mISDN channel driver (chan_misdn) changes
3026 ----------------------------------------
3027 * Added display_connected parameter to misdn.conf to put a display string
3028 in the CONNECT message containing the connected name and/or number if
3029 the presentation setting permits it.
3030 * Added display_setup parameter to misdn.conf to put a display string
3031 in the SETUP message containing the caller name and/or number if the
3032 presentation setting permits it.
3033 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3034 indicate the dialplan settings are to be obtained from the asterisk
3036 * Made misdn.conf parameter callerid accept the "name" <number> format
3037 used by the rest of the system.
3038 * Made use the nationalprefix and internationalprefix misdn.conf
3039 parameters to prefix any received number from the ISDN link if that
3040 number has the corresponding Type-Of-Number. NOTE: This includes
3041 comparing the incoming call's dialed number against the MSN list.
3042 * Added the following new parameters: unknownprefix, netspecificprefix,
3043 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3044 received number from the ISDN link if that number has the corresponding
3046 * Added new dialplan application misdn_command which permits controlling
3047 the CCBS/CCNR functionality.
3048 * Added new dialplan function mISDN_CC which permits retrieval of various
3049 values from an active call completion record.
3050 * For PTP, you should manually send the COLR of the redirected-to party
3051 for an incomming redirected call if the incoming call could experience
3052 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3053 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3054 if the REDIRECTING(from-num) is not empty.
3055 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3056 option on all of the REDIRECTING statements before dialing the
3057 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3058 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3059 redirecting-to presentation (COLR) when it becomes available.
3060 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3063 thirdparty mISDN enhancements
3064 -----------------------------
3065 mISDN has been modified by Digium, Inc. to greatly expand facility message
3067 * Enhanced COLP support for call diversion and transfer.
3068 * CCBS/CCNR support.
3070 The latest modified mISDN v1.1.x based version is available at:
3071 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3072 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3074 Tagged versions of the modified mISDN code are available under:
3075 http://svn.digium.com/svn/thirdparty/mISDN/tags
3076 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3078 libpri channel driver (chan_dahdi) DAHDI changes
3079 -------------------------------------------
3080 * The channel variable PRIREDIRECTREASON is now just a status variable
3081 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3082 to read and alter the reason.
3083 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3084 redirected-to party for an incomming redirected call if the incoming call
3085 could experience further redirects. Just set the
3086 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3087 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3089 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3090 use the inhibit(i) option on all of the REDIRECTING statements before
3091 dialing the redirected-to party. You still have to set the
3092 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3093 will update the redirecting-to presentation (COLR) when it becomes available.
3094 * Added the ability to ignore calls that are not in a Multiple Subscriber
3095 Number (MSN) list for PTMP CPE interfaces.
3096 * Added dynamic range compression support for dahdi channels. It is
3097 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3098 * Added support for ISDN calling and called subaddress with partial support
3099 for connected line subaddress.
3100 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3101 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3102 to transfer a held call on disconnect similar to an analog phone.
3103 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3104 Will reroute/deflect an outgoing call when receive the message.
3105 Can use the DAHDISendCallreroutingFacility to send the message for the
3107 * Added standard location to add options to chan_dahdi dialing:
3108 Dial(DAHDI/g1[/extension[/options]])
3111 R Reverse charging indication
3112 * Added Reverse Charging Indication (Collect calls) send/receive option.
3113 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3114 Dial(DAHDI/g1/extension/R)
3115 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3116 (requires latest LibPRI)
3117 * Added ability to send/receive keypad digits in the SETUP message.
3118 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3119 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3120 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3121 (requires latest LibPRI)
3122 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3123 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3124 back into the same interface. Tromboned calls happen because of call routing,
3125 call deflection, call forwarding, and call transfer.
3126 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3127 * Added the ability to support call waiting calls. (The SETUP has no B channel
3129 * Added Malicious Call ID (MCID) event to the AMI call event class.
3130 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3132 Asterisk Manager Interface
3133 --------------------------
3134 * The Hangup action now accepts a Cause header which may be used to
3135 set the channel's hangup cause.
3136 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3137 to specify a separate .pem file to hold a private key. By default sslcert
3138 is used to hold both the public and private key.
3139 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3140 for options containing the 'tls' prefix. For example, 'sslenable' is now
3141 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3142 across all .conf files. All affected sample.conf files have been modified to
3143 reflect this change. Previous options such as 'sslenable' still work,
3144 but options with the 'tls' prefix are preferred.
3145 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3146 in a channel. (res_mutestream.so)
3147 * The configuration file manager.conf now supports a channelvars option, which
3148 specifies a list of channel variables to include in each channel-oriented
3150 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3151 and ExtraPriority to allow redirecting the second channel to a different
3152 location than the first.
3153 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3155 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3156 in a MixMonitor recording.
3157 * The 'iax2 show peers' output is now similar to the expected output of
3159 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3161 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3162 AOC-E messages on a channel.
3163 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3164 conform more closely to similar events.
3165 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3167 * Added optional parkinglot variable for park command.
3168 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3169 if CallerIDNum and CallerIDName headers are also present.
3171 Channel Event Logging
3172 ---------------------
3173 * A new interface, CEL, is introduced here. CEL logs single events, much like
3174 the AMI, but it differs from the AMI in that it logs to db backends much
3175 like CDR does; is based on the event subsystem introduced by Russell, and
3176 can share in all its benefits; allows multiple backends to operate like CDR;
3177 is specialized to event data that would be of concern to billing sytems,
3178 like CDR. Backends for logging and accounting calls have been produced,
3179 but a new CDR backend is still in development.
3183 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3184 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3185 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3186 * Multiple files and formats can now be specified in cdr_custom.conf.
3187 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3188 See configs/cdr_syslog.conf.sample for more information.
3189 * A 'sequence' field has been added to CDRs which can be combined with
3190 linkedid or uniqueid to uniquely identify a CDR.
3191 * Handling of billsec and duration field has changed. If your table definition
3192 specifies those fields as float,double or similar they will now be logged with
3193 microsecond accuracy instead of a whole integer.
3195 Calendaring for Asterisk
3196 ------------------------
3197 * A new set of modules were added supporing calendar integration with Asterisk.
3198 Dialplan functions for reading from and writing to calendars are included,
3199 as well as the ability to execute dialplan logic upon calendar event notifications.
3200 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3201 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3202 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
3203 2003 support does not support forms-based authentication).
3205 Call Completion Supplementary Services for Asterisk
3206 ---------------------------------------------------
3207 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
3208 DAHDI/ISDN supports call completion for the following switch types:
3209 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
3210 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
3212 Multicast RTP Support
3213 ---------------------
3214 * A new RTP engine and channel driver have been added which supports Multicast RTP.
3215 The channel driver can be used with the Page application to perform multicast RTP
3216 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
3217 Type can be either basic or linksys.
3218 Destination is the IP address and port for the RTP packets.
3219 Control address is specific to the linksys type and is used for sending the control
3220 packets unique to them.
3222 Security Events Framework
3223 -------------------------
3224 * Asterisk has a new C API for reporting security events. The module res_security_log
3225 sends these events to the "security" logger level. Currently, AMI is the only
3226 Asterisk component that reports security events. However, SIP support will be
3227 coming soon. For more information on the security events framework, see the
3228 "Asterisk Security Framework" section of the Asterisk wiki at
3229 https://wiki.asterisk.org/wiki/x/wgBQ
3230 * SIP support was added in Asterisk 10
3231 * This API now supports IPv6 addresses
3235 * A technology independent fax frontend (res_fax) has been added to Asterisk.
3236 * A spandsp based fax backend (res_fax_spandsp) has been added.
3237 * The app_fax module has been deprecated in favor of the res_fax module and
3238 the new res_fax_spandsp backend.
3239 * The SendFAX and ReceiveFAX applications now send their log messages to a
3240 'fax' logger level, instead of to the generic logger levels. To see these
3241 messages, the system's logger.conf file will need to direct the 'fax' logger
3242 level to one or more destinations; the logger.conf.sample file includes an
3243 example of how to do this. Note that if the 'fax' logger level is *not*
3244 directed to at least one destination, log messages generated by these
3245 applications will be lost, and that if the 'fax' logger level is directed to
3246 the console, the 'core set verbose' and 'core set debug' CLI commands will
3247 have no effect on whether the messages appear on the console or not.
3251 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
3252 Now, in order to enable transmitting silence during record the transmit_silence
3253 option should be used. transmit_silence_during_record remains a valid option, but
3254 defaults to the behavior of the transmit_silence option.
3255 * Addition of the Unit Test Framework API for managing registration and execution
3256 of unit tests with the purpose of verifying the operation of C functions.
3257 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
3258 XMPP text messages to the remote JID.
3259 * Modules.conf has a new option - "require" - that marks a module as critical for
3260 the execution of Asterisk.
3261 If one of the required modules fail to load, Asterisk will exit with a return
3263 * An 'X' option has been added to the asterisk application which enables #exec support.
3264 This allows #exec to be used in asterisk.conf.
3265 * jabber.conf supports a new option auth_policy that toggles auto user registration.
3266 * A new lockconfdir option has been added to asterisk.conf to protect the
3267 configuration directory (/etc/asterisk by default) during reloads.
3268 * The parkeddynamic option has been added to features.conf to enable the creation
3269 of dynamic parkinglots.
3270 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
3271 the reportalarms config option.
3272 * chan_dahdi supports dialing configuring and dialing by device file name.
3273 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3274 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3275 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3276 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3277 Handy for the above name-based syntax as it does not depend on
3278 initialization order.
3279 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3280 significant increase in performance (about 3X) for installations using this switchtype.
3281 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3282 AIS. For more information, please see the Distributed Device State section of the
3283 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3284 * The addition of G.719 pass-through support.
3285 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3286 during device configuration.
3287 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3288 have less than 3 lines on the LCD.
3289 * Realtime now supports database failover. See the sample extconfig.conf for details.
3290 * The addition of improved translation path building for wideband codecs. Sample
3291 rate changes during translation are now avoided unless absolutely necessary.
3292 * The addition of the res_stun_monitor module for monitoring and reacting to network
3293 changes while behind a NAT.
3294 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3295 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3296 These allow support for any Administration. Default is AT&T values.
3300 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3301 optionally accept a filename, to apply the setting only to the code generated from
3302 that source file when Asterisk was built. However, there are some modules in Asterisk
3303 that are composed of multiple source files, so this did not result in the behavior
3304 that users expected. In this version, 'core set debug' and 'core set verbose'
3305 can optionally accept *module* names instead (with or without the .so extension),
3306 which applies the setting to the entire module specified, regardless of which source
3307 files it was built from.
3308 * New 'manager show settings' command showing the current settings loaded from
3310 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3311 the channel hangup request to all channels.
3312 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3314 ------------------------------------------------------------------------------
3315 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3316 ------------------------------------------------------------------------------
3320 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3321 Snom phones use this for call pickup of extensions that the phone is
3323 * Added support for setting the domain in the URI for caller of an
3324 outbound call by using the SIPFROMDOMAIN channel variable.
3325 * Added a new configuration option "remotesecret" for authentication to
3326 remote services. For backwards compatibility, "secret" still has the
3327 same function as before, but now you can configure both a remote secret and a
3328 local secret for mutual authentication.
3329 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3330 the sound will be played to the target of an attended transfer
3331 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3332 finer control over how many peers Asterisk will qualify and the gap between them
3333 when all peers need to be qualified at the same time.
3334 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3335 (either globally or for a specific peer), chan_sip will treat any SDP data
3336 it receives as new data and update the media stream accordingly. By
3337 default, Asterisk will only modify the media stream if the SDP session
3338 version received is different from the current SDP session version. This
3339 option is required to interoperate with devices that have non-standard SDP
3340 session version implementations (observed with Microsoft OCS). This option
3341 is disabled by default.
3342 * The parsing of register => lines in sip.conf has been modified to allow a port
3343 to be present in the "user" portion. Please see the sip.conf.sample file for more
3345 * Added support for subscribing to MWI on a remote server and making the status available
3346 as a mailbox. Please see the sip.conf.sample file for more information.
3347 * Added a function to remove SIP headers added in the dialplan before the
3348 first INVITE is generated - SIPRemoveHeader()
3349 * Channel variables set with setvar= in a device configuration is now
3350 set both for inbound and outbound calls.
3351 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3355 * Added immediate option to iax.conf
3356 * Added forceencryption option to iax.conf
3357 * Added Encryption and Trunk status to manager command "iaxpeers"
3361 * The configuration file now holds separate sections for devices and lines.
3362 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3367 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3368 support for LibOpenR2. http://www.libopenr2.org/
3369 * The UK option waitfordialtone has been added for use with BT analog
3371 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3372 is used in conjunction with the 'faxdetect' configuration option. When
3373 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3374 switch to the configured faxbuffers policy. For example, to use 6 buffers
3375 and a 'full' buffer policy for a fax transmission, add:
3377 The faxbuffers configuration will be in affect until the call is torn down.
3378 * Added service message support for 4ESS/5ESS switches.
3382 * For DAHDI channels, the CHANNEL() dialplan function now
3383 supports changing the channel's buffer policy (for the current
3384 call only), using this syntax:
3386 exten => s,n,Set(CHANNEL(buffers)=6,full)
3388 This would change the channel to the 'full' buffer policy and
3389 6 (six) buffers. Possible options for this setting are the same
3390 as those in chan_dahdi.conf.
3391 * Added a new dialplan function, CURLOPT, which permits setting various
3392 options that may be useful with the CURL dialplan function, such as
3393 cookies, proxies, connection timeouts, passwords, etc.
3394 * Permit the syntax and synopsis fields of the corresponding dialplan
3395 functions to be individually set from func_odbc.conf.
3396 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3397 * func_odbc now may specify an insert query to execute, when the write query
3398 affects 0 rows (usually indicating that no such row exists).
3399 * Added a new dialplan function, LISTFILTER, which permits removing elements
3400 from a set list, by name. Uses the same general syntax as the existing CUT
3401 and FIELDQTY dialplan functions, which also manage lists.
3402 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3403 obtaining realtime data from the dialplan.
3404 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3405 a subroutine when using the GoSub() and Return() applications.
3406 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3407 of "core show function AUDIOHOOK_INHERIT" from the CLI
3408 * Added AES_ENCRYPT. For information on its use, please see the output
3409 of "core show function AES_ENCRYPT" from the CLI
3410 * Added AES_DECRYPT. For information on its use, please see the output
3411 of "core show function AES_DECRYPT" from the CLI
3412 * func_odbc now supports database transactions across multiple queries.
3416 * Scheduled meetme conferences may now have their end times extended by
3418 * app_authenticate now gives the ability to select a prompt other than
3420 * app_directory now pays attention to the searchcontexts setting in
3421 voicemail.conf and will look through all contexts, if no context is
3422 specified in the initial argument.
3423 * A new application, Originate, has been introduced, that allows asynchronous
3424 call origination from the dialplan.
3425 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3426 in addition to the setting in the "general" context.
3427 * Added ConfBridge dialplan application which does conference bridges without
3428 DAHDI. For information on its use, please see the output of
3429 "core show application ConfBridge" from the CLI.
3433 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3434 operation to the AMI Redirect action.
3435 * extensions.conf now allows you to use keyword "same" to define an extension
3436 without actually specifying an extension. It uses exactly the same pattern
3437 as previously used on the last "exten" line. For example:
3438 exten => 123,1,NoOp(something)
3439 same => n,SomethingElse()
3440 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3441 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3442 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3443 by the new clialiases module. See cli_aliases.conf.sample file.
3444 * Times within timespecs are now accurate down to the minute. This is a change
3445 from historical Asterisk, which only provided timespecs rounded to the nearest
3446 even (read: evenly divisible by 2) minute mark.
3447 * The realtime switch now supports an option flag, 'p', which disables searches for
3449 * In addition to a time range and date range, timespecs now accept a 5th optional
3450 argument, timezone. This allows you to perform time checks on alternate
3451 timezones, especially if those daylight savings time ranges vary from your
3452 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3454 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3455 give you the correct output for an asterisk box behind nat. It will give you the
3456 externhost and localnet settings.
3457 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3458 can connect calls in passthrough mode, as well as record and play back files.
3459 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3460 using pickupsound and pickupfailsound in features.conf.
3461 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3462 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3463 instead of the /var/run/asterisk.pid where it used to be. This will make
3464 installs as non-root easier to manage.
3469 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3470 be written; they will no longer be explicitly written.
3472 Asterisk Manager Interface
3473 --------------------------
3474 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3475 a non-empty value) in your request. If you do this, any pending AMI events will
3476 *not* be included in the response to your request as they would normally, but
3477 will be left in the event queue for the next request you make to retrieve. For
3478 some applications, this will allow you to guarantee that you will only see
3479 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3480 To know whether the Asterisk server supports this header or not, your client can
3481 inspect the first response back from the server to see if it includes this header:
3483 Pragma: SuppressEvents
3485 If this is included, the server supports event suppression.
3487 * Added 4 new Actions to list skinny device(s) and line(s)
3493 LDAP Schema File Additions
3494 --------------------------
3495 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3496 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3498 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3499 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3500 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3501 * Removed redundant IPaddr (there's already IPAddress)
3502 - Gives more configuration Flags for SIP-Users available (tested)
3503 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3504 without extensibleObject (which really should be the last resort); gives
3505 also additional possibilities for LDAP-filter
3507 ------------------------------------------------------------------------------
projects / asterisk/asterisk.git / blob