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 13 to Asterisk 14 --------------------
13 ------------------------------------------------------------------------------
17 ------------------------------------------------------------------------------
18 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
19 ------------------------------------------------------------------------------
24 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
25 the focus of development for this release of Asterisk was on improving the
26 usability and features developed in the previous Standard release, Asterisk 12.
27 Beyond a general refinement of end user features, development focussed heavily
28 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
29 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
32 * Asterisk security events are now provided via AMI, allowing end users to
33 monitor their Asterisk system in real time for security related issues.
34 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
35 * Reception/transmission of out of call text messages using any supported
36 channel driver/protocol stack through ARI.
37 * Resource List Server support in the PJSIP stack, providing subscriptions to
38 lists of resources and batched delivery of NOTIFY requests.
39 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
42 It is important to note that Asterisk 13 is built on the architecture developed
43 during the previous Standard release, Asterisk 12. Users upgrading to
44 Asterisk 13 should read about the new features in Asterisk 12 later in this file
45 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
46 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
47 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
48 on AMI, CDRs, and CEL on the Asterisk wiki:
49 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
50 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
51 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
53 Many new featuers in Asterisk 13 were introduced in point releases of
54 Asterisk 12. Following this section - which documents the changes from all
55 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
56 that were introduced in the point releases of Asterisk 12, as they are also
57 included in Asterisk 13.
59 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
60 delivered with this release.
65 * Sample config files have been moved from configs/ to a sub-folder of that
68 * The menuselect utility has been pulled into the Asterisk repository. As a
69 result, the libxml2 development library is now a required dependency for
72 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
73 counted objects will emit additional debug information to the refs log file
74 located in the standard Asterisk log file directory. This log file is useful
75 in tracking down object leaks and other reference counting issues. Prior to
76 this version, this option was only available by modifying the source code
77 directly. This change also includes a new script, refcounter.py, in the
78 contrib folder that will process the refs log file. Note that this replaces
79 the refcounter utility that could be built from the utils directory.
87 * This module was deprecated and has been removed. Users of app_dahdibarge
88 should use ChanSpy instead.
92 * New options to play a beep when starting a recording and stopping a recording
93 have been added. The option "p" will play a beep to the channel that starts
94 the recording. The option "P" will play a beep to the channel that stops the
99 * This module was deprecated and has been removed. Users of app_readfile
100 should use func_env's FILE function instead.
104 * The 'say' family of dialplan applications now support the Japanese
105 language. The 'language' parameter in say.conf now recognizes a setting of
106 'ja', which will enable Japanese language specific mechanisms for playing
107 back numbers, dates, and other items.
111 * This module was deprecated and has been removed. Users of app_saycountpl
112 should use the Say family of applications.
116 * The SetMusicOnHold dialplan application was deprecated and has been removed.
117 Users of the application should use the CHANNEL function's musicclass
122 * The WaitMusicOnHold dialplan application was deprecated and has been
123 removed. Users of the application should use MusicOnHold with a duration
128 * VoiceMail and VoiceMailMain now support the Japanese language. The
129 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
130 which will enable prompts to be played back using a Japanese grammatical
131 structure. Additional prompts are necessary for this functionality,
133 - jb-arimasu: there is
134 - jb-arimasen: there is not
135 - jb-oshitekudasai: please press
141 * Add the ability to specify multiple email addresses in configuration,
150 * This module was deprecated and has been removed. Users of cdr_sqlite
151 should use cdr_sqlite3_custom.
155 * Added the ability to support PostgreSQL application_name on connections.
156 This allows PostgreSQL to display the configured name in the
157 pg_stat_activity view and CSV log entries. This setting is configurable
158 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
166 * Added the ability to support PostgreSQL application_name on connections.
167 This allows PostgreSQL to display the configured name in the
168 pg_stat_activity view and CSV log entries. This setting is configurable
169 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
177 * SS7 support now requires libss7 v2.0 or later.
179 * Added SS7 support for connected line and redirecting.
181 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
184 * Added several SS7 config option parameters described in
185 chan_dahdi.conf.sample.
189 * This module was deprecated and has been removed. Users of chan_gtalk
190 should use chan_motif.
194 * This module was deprecated and has been removed. Users of chan_h323
195 should use chan_ooh323.
199 * This module was deprecated and has been removed. Users of chan_jingle
200 should use chan_motif.
204 * The SIPPEER dialplan function no longer supports using a colon as a
205 delimiter for parameters. The parameters for the function should be
206 delimited using a comma.
208 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
209 of the function should use the CHANNEL function instead.
217 * Added functional peeraccount support. Except for Queue, the
218 accountcode propagation is now consistently propagated to outgoing
219 channels before dialing. The channel accountcode can change from its
220 original non-empty value on channel creation for the following specific
221 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
222 originate method that can specify an accountcode value. Three, the
223 calling channel propagates its peeraccount or accountcode to the
224 outgoing channel's accountcode before dialing. The change has two
225 visible effects. One, local channels now cross accountcode and
226 peeraccount across the special bridge between the ;1 and ;2 channels
227 just like channels between normal bridges. Two, the
228 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
229 set the accountcode on the outgoing channel(s).
231 For Queue, an outgoing channel's non-empty accountcode will not change
232 unless explicitly set by CHANNEL(accountcode). The change has three
233 visible effects. One, local channels now cross accountcode and
234 peeraccount across the special bridge between the ;1 and ;2 channels
235 just like channels between normal bridges. Two, the queue member will
236 get an accountcode if it doesn't have one and one is available from the
237 calling channel's peeraccount. Three, accountcode propagation includes
238 local channel members where the accountcodes are propagated early
239 enough to be available on the ;2 channel.
243 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
244 These events are emitted whenever a device state or presence state change
245 occurs. The events are controlled by res_manager_device_state.so and
246 res_manager_presence_state.so. If the high frequency of these events is
247 problematic for you, do not load these modules.
249 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
250 work in basically the same way as the 'dialplan add extension' and
251 'dialplan remove extension' CLI commands respectively.
253 * New AMI action LoggerRotate reloads and rotates logger in the same manner
254 as CLI command 'logger rotate'
256 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
257 functionality of CLI commands 'fax show sessions', 'fax show session',
258 and fax show stats' respectively.
260 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
261 enable manager control over PRI debugging levels and file output.
263 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
264 endpoint as long as a default outbound endpoint is set. This also applies
265 to the equivalent CLI command (pjsip send notify)
267 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
268 that give information on Asterisk's attempts to qualify the endpoint.
272 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
273 and BRIDGE_EXIT events.
277 * Channel variables are now substituted in arguments passed to applications
278 run by using dynamic features.
282 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
283 Enabling PFS is attempted by default, and is dependent on the configuration
284 of the module using TLS.
285 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
286 specify a ECDHE cipher suite in sip.conf, for example:
287 tlscipher=AES128-SHA:DES-CBC3-SHA
288 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
289 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
290 default dh2048.pem - see
291 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
292 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
293 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
294 Consider re-ordering your cipher suites in the respective configuration
296 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
297 will use PFS when offered by the client. Clients which do not offer PFS
298 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
306 * The JACK_HOOK function now supports audio with a sample rate higher than
315 * Added the ability to support PostgreSQL application_name on connections.
316 This allows PostgreSQL to display the configured name in the
317 pg_stat_activity view and CSV log entries. This setting is configurable
318 for res_config_pgsql via the dbappname configuration setting in
321 res_pjsip_outbound_publish
323 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
324 PUBLISH requests for specific event packages to another SIP User Agent.
328 * The publish/subscribe core module has been updated to support RFC 4662
329 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
330 Resource lists are configured in pjsip.conf under a new object type,
331 resource_list. Resource lists can contain either message-summary or presence
332 events, and can be composed of specific resources that provide the event or
333 other resource lists.
335 * Inbound publication support is provided by a new object, inbound-publication.
336 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
337 resource. Which events are accepted is constructed dynamically; see
338 res_pjsip_publish_asterisk for more information.
340 res_pjsip_publish_asterisk
342 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
343 Asterisk information to other Asterisk servers. This module is intended only
344 for Asterisk to Asterisk exchanges of information. Currently, this includes
345 both mailbox state and device state information.
348 ------------------------------------------------------------------------------
349 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
350 ------------------------------------------------------------------------------
354 * Stored recordings now support a new operation, copy. This will take an
355 existing stored recording and copy it to a new location in the recordings
358 * LiveRecording objects now have three additional fields that can be reported
359 in a RecordingFinished ARI event:
360 - total_duration: the duration of the recording
361 - talking_duration: optional. The duration of talking detected in the
362 recording. This is only available if max_silence_seconds was specified
363 when the recording was started.
364 - silence_duration: optional. The duration of silence detected in the
365 recording. This is only available if max_silence_seconds was specified
366 when the recording was started.
367 Note that all duration values are reported in seconds.
369 * Users of ARI can now send and receive out of call text messages. Messages
370 can be sent directly to a particular endpoint, or can be sent to the
371 endpoints resource directly and inferred from the URI scheme. Text
372 messages are passed to ARI clients as TextMessageReceived events. ARI
373 clients can choose to receive text messages by subscribing to the particular
374 endpoint technology or endpoints that they are interested in.
376 * The applications resource now supports subscriptions to all endpoints of
377 a particular channel technology. For example, subscribing to an eventSource
378 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
382 * The endpoint configuration object now supports 'accountcode'. Any channel
383 created for an endpoint with this setting will have its accountcode set
384 to the specified value.
388 * A new module, res_hep_rtcp, has been added that will forward RTCP call
389 statistics to a HEP capture server. See res_hep for more information.
393 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
394 unconditionally inhereted through masquerades. As a side benefit, more
395 than one audiohook of a given type may persist through a masquerade now.
397 ------------------------------------------------------------------------------
398 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
399 ------------------------------------------------------------------------------
403 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
404 connect with an incoming caller after being alerted to the presence
405 of the incoming caller. The most likely reason this would happen is
406 the agent did not acknowledge the call in time.
410 * New events have been added for the TALK_DETECT function. When the function
411 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
412 emitted to connected AMI clients indicating the start/stop of talking on
417 * New event models have been aded for the TALK_DETECT function. When the
418 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
419 events will be emitted to connected WebSockets subscribed to the channel,
420 indicating the start/stop of talking on the channel.
424 * A new function, TALK_DETECT, has been added. When set on a channel, this
425 fucntion causes events indicating the starting/stoping of talking on said
426 channel to be emitted to both AMI and ARI clients.
428 ------------------------------------------------------------------------------
429 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
430 ------------------------------------------------------------------------------
434 * A new Playback URI 'tone' has been added. Tones are specified either as
435 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
436 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
437 URIs in that they must be stopped manually and will continue to occupy
438 a channel's ARI control queue until they are stopped. They also can not
439 be rewound or fastforwarded.
441 * User events can now be generated from ARI. Events can be signalled with
442 arbitrary json variables, and include one or more of channel, bridge, or
443 endpoint snapshots. An application must be specified which will receive
444 the event message (other applications can subscribe to it). The message
445 will also be delivered via AMI provided a channel is attached. Dialplan
446 generated user event messages are still transmitted via the channel, and
447 will only be received by a stasis application they are attached to or if
448 the channel is subscribed to.
452 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
453 fields for prohibited callingpres information. Values are legacy, no, and
454 yes. By default, legacy is used.
455 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
456 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
457 headers are appended to outbound SIP messages just as they are with
458 allowed callingpres values, but data about the remote party's identity is
460 When sendrpid=rpid, only the remote party's domain is anonymized.
461 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
462 headers are not sent.
463 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
464 party information in tact even for prohibited callingpres information.
465 In the case of PAI, a Privacy: id header will be appended for prohibited
466 calling information to communicate that the private information should
467 not be relayed to untrusted parties.
471 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
472 which can be used to announce the parked call's location to an arbitrary
473 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
474 parties in a one to one bridge, 'TimeoutChannel' is treated as having
475 parked 'Channel' like with the Park Call DTMF feature and will receive
476 announcements prior to being hung up.
478 ------------------------------------------------------------------------------
479 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
480 ------------------------------------------------------------------------------
484 * Record application now has an option 'o' which allows 0 to act as an exit
485 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
488 --------------------------
489 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
490 as the chanprefix parameter if the 'u' option is specified.
493 --------------------------
494 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
495 conference user menus.
497 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
498 menus, bridge settings, and user settings that have been applied by the
499 CONFBRIDGE dialplan function.
501 * The ConfBridge dialplan application now sets a channel variable,
502 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
503 how a channel exited the conference.
505 * Added conference user option 'announce_join_leave_review'. This option
506 implies 'announce_join_leave' with the added effect that the user will
507 be asked if they want to confirm or re-record the recording of their
508 name when entering the conference
511 --------------------------
512 * At exit, the Directory application now sets a channel variable
513 DIRECTORY_RESULT to one of the following based on the reason for exiting:
514 OPERATOR user requested operator by pressing '0' for operator
515 ASSISTANT user requested assistant by pressing '*' for assistant
516 TIMEOUT user pressed nothing and Directory stopped waiting
517 HANGUP user's channel hung up
518 SELECTED user selected a user from the directory and is routed
519 USEREXIT user pressed '#' from the selection prompt to exit
520 FAILED directory failed in a way that wasn't accounted for. Dang.
524 * Monitor() - A new option, B(), has been added that will turn on a periodic
525 beep while the call is being recorded.
528 --------------------------
529 * MusicOnHold streams (all modes other than "files") now support wide band
533 --------------------------
534 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
535 and for the channel executing Page respectively.
538 --------------------------
539 * PickupChan now accepts channel uniqueids of channels to pickup.
542 --------------------------
543 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
544 to 'true' (case insensitive), then any Say application (SayNumber,
545 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
546 anticipate DTMF. If DTMF is received, these applications will behave like
547 the background application and jump to the received extension once a match
548 is established or after a short period of inactivity.
551 -------------------------
552 * A new function, MIXMONITOR, has been added to allow access to individual
553 instances of MixMonitor on a channel.
555 * A new option, B(), has been added that will turn on a periodic beep while the
556 call is being recorded.
560 -------------------------
563 -------------------------
564 * TEL URI support for inbound INVITE requests has been added. chan_sip will
565 now handle TEL schemes in the Request and From URIs. The phone-context in
566 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
571 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
572 the new AST_SORCERY diaplan function.
574 * Core Show Locks output now includes Thread/LWP ID if the platform
575 supports this feature.
577 * New "logger add channel" and "logger remove channel" CLI commands have
578 been added to allow creation and deletion of dynamic logger channels
579 without configuration changes. These dynamic logger channels will only
580 exist until the next restart of asterisk.
584 * The live recording object on recording events now contains a target_uri
585 field which contains the URI of what is being recorded.
587 * The bridge type used when creating a bridge is now a comma separated list of
588 bridge properties. Valid options are: mixing, holding, dtmf_events, and
591 * A channelId can now be provided when creating a channel, either in the
592 uri (POST channels/my-channel-id) or as query parameter. A local channel
593 will suffix the second channel id with ';2' unless provided as query
594 parameter otherChannelId.
596 * A bridgeId can now be provided when creating a bridge, either in the uri
597 (POST bridges/my-bridge-id) or as a query parameter.
599 * A playbackId can be provided when starting a playback, either in the uri
600 (POST channels/my-channel-id/play/my-playback-id /
601 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
603 * A snoop channel can be started with a snoopId, in the uri or query.
607 * Originate now takes optional parameters ChannelId and OtherChannelId,
608 used to set the UniqueId on creation. The other id is assigned to the
609 second channel when dialing LOCAL, or defaults to appending ;2 if only
610 the single Id is given.
612 * The Mixmonitor action now has a "Command" header that can be used to
613 indicate a post-process command to run once recording finishes.
617 * A new set of Alembic scripts has been added for CDR tables. This will create
618 a 'cdr' table with the default schema that Asterisk expects.
623 * A new function was added: PERIODIC_HOOK. This allows running a periodic
624 dialplan hook on a channel. Any audio generated by this hook will be
625 injected into the call.
633 * A new module, res_hep, has been added, that acts as a generic packet
634 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
635 It can be configured via hep.conf. Other modules can use res_hep to send
636 message traffic to a HEP capture server.
640 * A new module, res_hep_pjsip, has been added that will forward PJSIP
641 message traffic to a HEP capture server. See res_hep for more
646 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
647 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
649 * Added the following new CLI commands:
650 - "pjsip show contacts" - list all current PJSIP contacts.
651 - "pjsip show contact" - show specific information about a current PJSIP
653 - "pjsip show channel" - show detailed information about a PJSIP channel.
657 * A new module, res_pjsip_multihomed handles situations where the system
658 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
659 determines which interface should be used during message sending.
661 res_pjsip_pidf_digium_body_supplement
663 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
664 request body formatting for presence support in Digium phones.
666 res_pjsip_send_to_voicemail
668 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
669 particular headers to transfer a PJSIP channel directly to a particular
670 extension that has VoiceMail. This is intended to be used with Digium
671 phones that support this feature.
673 res_pjsip_outbound_registration
675 * A new CLI command has been added: "pjsip show registrations", which lists
676 all configured PJSIP registrations
679 ------------------------------------------------------------------------------
680 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
681 ------------------------------------------------------------------------------
685 * Added a new module that provides AMI control over MWI within Asterisk,
686 res_mwi_external_ami. Note that this module depends on res_mwi_external;
687 for more information on enabling this module, see res_mwi_external.
688 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
689 the MWIGet/MWIGetComplete events.
691 * The DialStatus field in the DialEnd event can now contain additional
692 statuses that convey how the dial operation terminated. This includes
693 ABORT, CONTINUE, and GOTO.
695 * AMI will now emit security events. A new class authorization has been
696 added in manager.conf for the security events, 'security'. The new events
698 - FailedACL - raised when a request violates an ACL check
699 - InvalidAccountID - raised when a request fails an authentication
700 check due to an invalid account ID
701 - SessionLimit - raised when a request fails due to exceeding the
702 number of allowed concurrent sessions for a service
703 - MemoryLimit - raised when a request fails due to an internal memory
705 - LoadAverageLimit - raised when a request fails because a configured
706 load average limit has been reached
707 - RequestNotAllowed - raised when a request is not allowed by
709 - AuthMethodNotAllowed - raised when a request used an authentication
710 method not allowed by the service
711 - RequestBadFormat - raised when a request is received with bad formatting
712 - SuccessfulAuth - raised when a request successfully authenticates
713 - UnexpectedAddress - raised when a request has a different source address
714 then what is expected for a session already in progress with a service
715 - ChallengeResponseFailed - raised when a request's attempt to authenticate
716 has been challenged, and the request failed the authentication challenge
717 - InvalidPassword - raised when a request provides an invalid password
718 during an authentication attempt
719 - ChallengeSent - raised when an Asterisk service send an authentication
720 challenge to a request
721 - InvalidTransport - raised when a request attempts to use a transport not
722 allowed by the Asterisk service
724 * Bridge related events now have two additional fields: BridgeName and
725 BridgeCreator. BridgeName is a descriptive name for the bridge;
726 BridgeCreator is the name of the entity that created the bridge. This
727 affects the following events: ConfbridgeStart, ConfbridgeEnd,
728 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
729 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
730 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
734 * The Bridge data model now contains the additional fields 'name' and
735 'creator'. The 'name' field conveys a descriptive name for the bridge;
736 the 'creator' field conveys the name of the entity that created the bridge.
737 This affects all responses to HTTP requests that return a Bridge data model
738 as well as all event derived data models that contain a Bridge data model.
739 The POST /bridges operation may now optionally specify a name to give to
740 the bridge being created.
742 * Added a new ARI resource 'mailboxes' which allows the creation and
743 modification of mailboxes managed by external MWI. Modules res_mwi_external
744 and res_stasis_mailbox must be enabled to use this resource. For more
745 information on external MWI control, see res_mwi_external.
747 * Added new events for externally initiated transfers. The event
748 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
749 of a bridge in the ARI controlled application to the dialplan; the
750 BridgeAttendedTransfer event is raised when a channel initiates an
751 attended transfer of a bridge in the ARI controlled application to the
754 * Channel variables may now be specified as a body parameter to the
755 POST /channels operation. The 'variables' key in the JSON is interpreted
756 as a sequence of key/value pairs that will be added to the created channel
757 as channel variables. Other parameters in the JSON body are treated as
758 query parameters of the same name.
762 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
763 automatically handled by the HTTP server if a request is received with a
764 Transfer-Encoding type of "chunked".
768 * Path support has been added with the 'support_path' option in registration
771 * A 'debug' option has been added to the globals section that will allow
772 sip messages to be logged.
774 * A 'set_var' option has been added to endpoints that will automatically
775 set the desired variable(s) on a channel created for that endpoint.
777 * Several new tables and columns have been added to the realtime schema for
778 the res_pjsip related modules. See the UPGRADE.txt notes for updating
783 * A new module, res_mwi_external, has been added to Asterisk. This module
784 acts as a base framework that other modules can build on top of to allow
785 an external system to control MWI within Asterisk. For implementations
786 that make use of res_mwi_external, see res_mwi_external_ami and
787 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
788 that may produce MWI themselves, such as app_voicemail. res_mwi_external
789 and other modules that depend on it cannot be built or loaded with
790 app_voicemail present.
794 * DNS functionality will now automatically be enabled if the system configured
795 nameservers can be retrieved. If the system configured nameservers can not be
796 retrieved the functionality will resort to using system resolution. Functionalty
797 such as SRV records and failover will not be available if system resolution
800 ------------------------------------------------------------------------------
801 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
802 ------------------------------------------------------------------------------
807 Asterisk 12 is a standard release of the Asterisk project. As such, the
808 focus of development for this release was on core architectural changes and
809 major new features. This includes:
810 * A more flexible bridging core based on the Bridging API
811 * A new internal message bus, Stasis
812 * Major standardization and consistency improvements to AMI
813 * Addition of the Asterisk RESTful Interface (ARI)
814 * A new SIP channel driver, chan_pjsip
815 In addition, as the vast majority of bridging in Asterisk was migrated to the
816 Bridging API used by ConfBridge, major changes were made to most of the
817 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
819 Specifications have been written for the affected interfaces. These
820 specifications are available on the Asterisk wiki:
821 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
822 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
823 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
825 It is *highly* recommended that anyone migrating to Asterisk 12 read the
826 information regarding its release both in this file and in the accompanying
827 UPGRADE.txt file. More detailed information on the major changes can be found
828 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
833 * Added build option DISABLE_INLINE. This option can be used to work around a
834 bug in gcc. For more information, see
835 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
837 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
838 the CHANNEL_TRACE build option were incompatible with the new bridging
841 * Asterisk now optionally uses libxslt to improve XML documentation generation
842 and maintainability. If libxslt is not available on the system, some XML
843 documentation will be incomplete.
845 * Asterisk now depends on libjansson. If a package of libjansson is not
846 available on your distro, please see http://www.digip.org/jansson/.
848 * Asterisk now depends on libuuid and, optionally, uriparser. It is
849 recommended that you install uriparser, even if it is optional.
851 * The new SIP stack and channel driver uses a particular version of PJSIP.
852 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
853 configuring and installing PJSIP for usage with Asterisk.
855 * Optional API was re-implemented to be more portable, and no longer requires
856 weak reference support from the compiler. The build option OPTIONAL_API may
857 be disabled to disable Optional API support.
864 * Along with AgentRequest, this application has been modified to be a
865 replacement for chan_agent. The act of a channel calling the AgentLogin
866 application places the channel into a pool of agents that can be
867 requested by the AgentRequest application. Note that this application, as
868 well as all other agent related functionality, is now provided by the
869 app_agent_pool module. See chan_agent and AgentRequest for more information.
871 * This application no longer performs agent authentication. If authentication
872 is desired, the dialplan needs to perform this function using the
873 Authenticate or VMAuthenticate application or through an AGI script before
876 * If this application is called and the agent is already logged in, the
877 dialplan will continue exection with the AGENT_STATUS channel variable set
878 to ALREADY_LOGGED_IN.
880 * The agents.conf schema has changed. Rather than specifying agents on a
881 single line in comma delineated fashion, each agent is defined in a separate
882 context. This allows agents to use the power of context templates in their
885 * A number of parameters from agents.conf have been removed. This includes
886 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
887 urlprefix, and savecallsin. These options were obsoleted by the move from
888 a channel driver model to the bridging/application model provided by
893 * A new application, this will request a logged in agent from the pool and
894 bridge the requested channel with the channel calling this application.
895 Logged in agents are those channels that called the AgentLogin application.
896 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
897 application will be set with an appropriate error value.
901 * This application has been removed. It was a holdover from when
902 AgentCallbackLogin was removed.
906 * Added support for additional Ademco DTMF signalling formats, including
907 Express 4+1, Express 4+2, High Speed and Super Fast.
909 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
910 call time, in milliseconds, to run the application.
912 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
913 maximum number of times to retry the call.
915 * Added a new configuration option answait. If set, the AlarmReceiver
916 application will wait the number of milliseconds specified by answait
917 after the channel has answered. Valid values range between 500
918 milliseconds and 10000 milliseconds.
920 * Added configuration option no_group_meta. If enabled, grouping of metadata
921 information in the AlarmReceiver log file will be skipped.
925 * It is now no longer possible to bypass updating the CDR on the channel
926 when answering. CDRs reflect the state of the channel and will always
927 reflect the time they were Answered.
931 * A new application in Asterisk, this will place the calling channel
932 into a holding bridge, optionally entertaining them with some form of
933 media. Channels participating in a holding bridge do not interact with
934 other channels in the same holding bridge. Optionally, however, a channel
935 may join as an announcer. Any media passed from an announcer channel is
936 played to all channels in the holding bridge. Channels leave a holding
937 bridge either when an optional timer expires, or via the ChannelRedirect
938 application or AMI Redirect action.
942 * All participants in a bridge can now be kicked out of a conference room
943 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
944 command, i.e., 'confbridge kick <conference> all'
946 * CLI output for the 'confbridge list' command has been improved. When
947 displaying information about a particular bridge, flags will now be shown
948 for the participating users indicating properties of that user.
950 * The ConfbridgeList event now contains the following fields: WaitMarked,
951 EndMarked, and Waiting. This displays additional properties about the
952 user's profile, as well as whether or not the user is waiting for a
953 Marked user to enter the conference.
955 * Added a new option for conference recording, record_file_append. If enabled,
956 when the recording is stopped and then re-started, the existing recording
957 will be used and appended to.
959 * ConfBridge now has the ability to set the language of announcements to the
960 conference. The language can be set on a bridge profile in confbridge.conf
961 or by the dialplan function CONFBRIDGE(bridge,language)=en.
965 * The channel variable CPLAYBACKSTATUS may now return the value
966 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
967 such as AMI. See the AMI action ControlPlayback for more information.
971 * Added the 'a' option, which allows the caller to enter in an additional
972 alias for the user in the directory. This option must be used in conjunction
973 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
974 specified in voicemail.conf.
978 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
979 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
980 containing the unique ID of the bridge that the channel happens to be in.
984 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
985 for more information.
987 * Variables are no longer purged from the original CDR. See the 'v' option for
990 * The 'A' option has been removed. The Answer time on a CDR is never updated
993 * The 'd' option has been removed. The disposition on a CDR is a function of
994 the state of the channel and cannot be altered.
996 * The 'D' option has been removed. Who the Party B is on a CDR is a function
997 of the state of the respective channels involved in the CDR and cannot be
1000 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1001 such that the start time and, if applicable, the answer time was updated.
1002 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1003 'r' option now triggers the Reset, setting the start time (and answer time
1004 if applicable) to the current time. Note that the 'a' option still sets
1005 the answer time to the current time if the channel was already answered.
1007 * The 's' option has been removed. A variable can be set on the original CDR
1008 if desired using the CDR function, and removed from a forked CDR using the
1011 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1012 longer applies in the CDR engine.
1014 * The 'v' option now prevents the copy of the variables from the original CDR
1015 to the forked CDR. Previously the variables were always copied but were
1016 removed from the original. This was changed as removing variables from a CDR
1017 can have unintended side effects - this option allows the user to prevent
1018 propagation of variables from the original to the forked without modifying
1023 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1024 function to a channel joining a conference. Some channel drivers that vary
1025 the number of audio samples in a voice frame will experience significant
1026 quality problems if a denoiser is attached to the channel; this option gives
1027 them the ability to remove the denoiser without having to unload func_speex.
1031 * The 'b' option now includes conferences as well as sounds played to the
1034 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1035 running during a transfer. If a MixMonitor is started on a channel,
1036 the MixMonitor will continue to record the audio passing through the
1037 channel even in the presence of transfers.
1041 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1044 * While the NoCDR application will prevent CDRs for a channel from being
1045 propagated to registered CDR backends, it will not prevent that data from
1046 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1047 function that enables CDRs on a channel will restore those records that have
1048 not yet been finalized.
1052 * The app_parkandannounce module has been removed. The application
1053 ParkAndAnnounce is now provided by the res_parking module. See the
1054 res_parking changes for more information.
1058 * Added queue available hint. The hint can be added to the dialplan using the
1059 following syntax: exten,hint,Queue:{queue_name}_avail
1060 For example, if the name of the queue is 'markq':
1061 exten => 8501,hint,Queue:markq_avail
1062 This will report 'InUse' if there are no logged in agents or no free agents.
1063 It will report 'Idle' when an agent is free.
1065 * Queues now support a hint for member paused state. The hint uses the form
1066 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1067 are the name of the queue and the name of the member to subscribe to,
1068 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1069 Members will show as In Use when paused.
1071 * The configuration options eventwhencalled and eventmemberstatus have been
1072 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1073 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1074 sent. The "Variable" fields will also no longer exist on the Agent* events.
1075 These events can be filtered out from a connected AMI client using the
1076 eventfilter setting in manager.conf.
1078 * The queue log now differentiates between blind and attended transfers. A
1079 blind transfer will result in a BLINDTRANSFER message with the destination
1080 context and extension. An attended transfer will result in an
1081 ATTENDEDTRANSFER message. This message will indicate the method by which
1082 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1083 for running an application on a bridge or channel, or "LINK" for linking
1084 two bridges together with local channels. The queue log will also now detect
1085 externally initiated blind and attended transfers and record the transfer
1088 * When performing queue pause/unpause on an interface without specifying an
1089 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1090 least one member of any queue exists for that interface.
1092 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1093 for realtime queue log entries.
1097 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1098 CDRs when they were previously disabled on a channel.
1100 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1101 backends occurs on an as-needed basis in order to preserve linkedid
1102 propagation and other needed behavior.
1106 * A new application, this is similar to SayAlpha except that it supports
1107 case sensitive playback of the specified characters. For example,
1108 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1112 * This application is deprecated in favor of CHANNEL(amaflags).
1116 * The SendDTMF application will now accept 'W' as valid input. This will cause
1117 the application to delay one second while streaming DTMF.
1121 * A new application in Asterisk 12, this hands control of the channel calling
1122 the application over to an external system. Currently, external systems
1123 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1127 * UserEvent will now handle duplicate keys by overwriting the previous value
1128 assigned to the key.
1130 * In addition to AMI, UserEvent invocations will now be distributed to any
1131 interested Stasis applications.
1135 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1136 system as mailbox@context. The rest of the system cannot add @default
1137 to mailbox identifiers for app_voicemail that do not specify a context
1138 any longer. It is a mailbox identifier format that should only be
1139 interpreted by app_voicemail.
1141 * The voicemail.conf configuration file now has an 'alias' configuration
1142 parameter for use with the Directory application. The voicemail realtime
1143 database table schema has also been updated with an 'alias' column.
1148 * Pass through support has been added for both VP8 and Opus.
1150 * Added format attribute negotiation for the Opus codec. Format attribute
1151 negotiation is provided by the res_format_attr_opus module.
1156 * Masquerades as an operation inside Asterisk have been effectively hidden
1157 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1158 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1159 dropping of frame/audio hooks, and other internal implementation details
1160 that users had to deal with. This fundamental change has large implications
1161 throughout the changes documented for this version. For more information
1162 about the new core architecture of Asterisk, please see the Asterisk wiki.
1164 * Multiple parties in a bridge may now be transferred. If a participant in a
1165 multi-party bridge initiates a blind transfer, a Local channel will be used
1166 to execute the dialplan location that the transferer sent the parties to. If
1167 a participant in a multi-party bridge initiates an attended transfer,
1168 several options are possible. If the attended transfer results in a transfer
1169 to an application, a Local channel is used. If the attended transfer results
1170 in a transfer to another channel, the resulting channels will be merged into
1173 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1174 driver specific. If the channel variable is set on the transferrer channel,
1175 the sound will be played to the target of an attended transfer.
1177 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1178 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1179 listed. Any more peers in the bridge will not be included in the list.
1180 BRIDGEPEER is not valid in holding bridges like parking since those channels
1181 do not talk to each other even though they are in a bridge.
1183 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1184 and will contain a value if the BRIDGEPEER's channel driver supports it.
1186 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1187 was responsible for an attended transfer in a similar fashion to
1190 * Modules using the Configuration Framework or Sorcery must have XML
1191 configuration documentation. This configuration documentation is included
1192 with the rest of Asterisk's XML documentation, and is accessible via CLI
1193 commands. See the CLI changes for more information.
1195 AMI (Asterisk Manager Interface)
1197 * Major changes were made to both the syntax as well as the semantics of the
1198 AMI protocol. In particular, AMI events have been substantially improved
1199 in this version of Asterisk. For more information, please see the AMI
1200 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1202 * AMI events that reference a particular channel or bridge will now always
1203 contain a standard set of fields. When multiple channels or bridges are
1204 referenced in an event, fields for at least some subset of the channels
1205 and bridges in the event will be prefixed with a descriptive name to avoid
1206 name collisions. See the AMI event documentation on the Asterisk wiki for
1209 * The CLI command 'manager show commands' no longer truncates command names
1210 longer than 15 characters and no longer shows authorization requirement
1211 for commands. 'manager show command' now displays the privileges needed
1212 for using a given manager command instead.
1214 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1215 peer in its response if the peer has a subscribe context set.
1217 * The SIPqualifypeer action now acknowledges the request once it has
1218 established that the request is against a known peer. It also issues a new
1219 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1221 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1222 specifies the duration of the digit to be played, in milliseconds.
1224 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1225 updates when changes occur instead of requiring the use of pollmailboxes.
1227 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1228 AMI client to manipulate audio currently being played back on a channel. The
1229 supported operations depend on the application being used to send audio to
1230 the channel. When the audio playback was initiated using the ControlPlayback
1231 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1232 stopped, restarted, reversed, or skipped forward. When initiated by other
1233 mechanisms (such as the Playback application), the audio can be stopped,
1234 reversed, or skipped forward.
1236 * Channel related events now contain a snapshot of channel state, adding new
1237 fields to many of these events.
1239 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1240 in a future release. Please use the common 'Exten' field instead.
1242 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1243 fields. The channel state fields will come before the body fields.
1245 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1246 'UnParkedCall' have changed significantly in the new res_parking module.
1248 The 'Channel' and 'From' headers are gone. For the channel that was parked
1249 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1250 has a number of fields associated with it. The old 'Channel' header relayed
1251 the same data as the new 'ParkeeChannel' header.
1253 The 'From' field was ambiguous and changed meaning depending on the event.
1254 for most of these, it was the name of the channel that parked the call
1255 (the 'Parker'). There is no longer a header that provides this channel name,
1256 however the 'ParkerDialString' will contain a dialstring to redial the
1257 device that parked the call.
1259 On UnParkedCall events, the 'From' header would instead represent the
1260 channel responsible for retrieving the parkee. It receives a channel
1261 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1264 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1266 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1267 fashion has changed the field names 'StartExten' and 'StopExten' to
1268 'StartSpace' and 'StopSpace' respectively.
1270 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1271 manager.conf has been removed.
1273 * Channel Variables conveyed with a channel no longer contain the name of the
1274 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1275 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1276 event, the various ChanVariable fields will contain a suffix that specifies
1277 which channel they correspond to.
1279 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1280 event always conveys the AMI event for a particular channel.
1282 * All 'Reload' events have been consolidated into a single event type. This
1283 event will always contain a Module field specifying the name of the module
1284 and a Status field denoting the result of the reload. All modules now issue
1285 this event when being reloaded.
1287 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1288 fail to receive this event due to being connected after modules have loaded.
1289 AMI connections that want to know when Asterisk is ready should listen for
1290 the 'FullyBooted' event.
1292 * app_fax now sends the same send fax/receive fax events as res_fax. The
1293 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1294 now the 'ReceiveFAX' event.
1296 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1297 'MusicOnHoldStop'. The sub type field has been removed.
1299 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1300 carrier for another protocol.
1302 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1303 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1304 to the specific channel. 'Both' may be specified to play a tone to both
1305 channels. The old 'yes' option is still accepted as a way of playing the
1306 tone to Channel2 only.
1308 * The AMI 'Status' response event to the AMI Status action replaces the
1309 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1310 indicate what bridge the channel is currently in.
1312 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1313 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1316 * The AMI events in app_queue have been made more consistent with each other.
1317 Events that reference channels (QueueCaller* and Agent*) will show
1318 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1319 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1321 * The 'MCID' AMI event now publishes a channel snapshot when available and
1322 its non-channel-snapshot parameters now use either the "MCallerID" or
1323 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1324 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1325 parameters in the channel snapshot.
1327 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1328 'AgentLogin' and 'AgentLogoff' respectively.
1330 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1331 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1333 * 'ChannelUpdate' events have been removed.
1335 * All AMI events now contain a 'SystemName' field, if available.
1337 * Local channel optimization is now conveyed in two events:
1338 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1339 when the Local channel driver begins attempting to optimize itself out of
1340 the media path; the End event is sent after the channel halves have
1341 successfully optimized themselves out of the media path.
1343 * Local channel information in events is now prefixed with 'LocalOne' and
1344 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1345 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1346 and 'LocalOptimizationEnd' events.
1348 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1349 account. When set in the general context, it will act as the default
1350 setting for defined accounts.
1352 * The 'BridgeAction' event was removed. It technically added no value, as the
1353 Bridge Action already receives confirmation of the bridge through a
1354 successful completion Event.
1356 * The 'BridgeExec' events were removed. These events duplicated the events that
1357 occur in the Briding API, and are conveyed now through BridgeCreate,
1358 BridgeEnter, and BridgeLeave events.
1360 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1361 previous versions. They now report all SR/RR packets sent/received, and
1362 have been restructured to better reflect the data sent in a SR/RR. In
1363 particular, the event structure now supports multiple report blocks.
1365 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1366 raised when a blind transfer/attended transfer completes successfully.
1367 They contain information about the transfer that just completed, including
1368 the location of the transfered channel.
1370 * Added a 'security' class to AMI which outputs the required fields for
1371 security messages similar to the log messages from res_security_log
1373 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1374 that describes the status value in a human readable string.
1376 CDR (Call Detail Records)
1378 * Significant changes have been made to the behavior of CDRs. The CDR engine
1379 was effectively rewritten and built on the Stasis message bus. For a full
1380 definition of CDR behavior in Asterisk 12, please read the specification
1381 on the Asterisk wiki (wiki.asterisk.org).
1383 * CDRs will now be created between all participants in a bridge. For each
1384 pair of channels in a bridge, a CDR is created to represent the path of
1385 communication between those two endpoints. This lets an end user choose who
1386 to bill for what during bridge operations with multiple parties.
1388 * The duration, billsec, start, answer, and end times now reflect the times
1389 associated with the current CDR for the channel, as opposed to a cumulative
1390 measurement of all CDRs for that channel.
1392 * When a CDR is dispatched, user defined CDR variables from both parties are
1393 included in the resulting CDR. If both parties have the same variable, only
1394 the Party A value is provided.
1396 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1397 information regarding the CDR engine is logged as verbose messages. This
1398 option should only be used if the behavior of the CDR engine needs to be
1401 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1402 normally configured in cdr.conf.
1404 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1405 specified, this command provides a summary of the channels with CDR
1406 information and their statistics. When {channel} is specified, it shows
1407 detailed information about all records associated with {channel}.
1409 CEL (Channel Event Logging)
1411 * CEL has undergone significant rework in Asterisk 12, and is now built on the
1412 Stasis message bus. Please see the specification for CEL on the Asterisk
1413 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
1416 * The 'extra' field of all CEL events that use it now consists of a JSON blob
1417 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
1419 * BLINDTRANSFER events now report the transferee bridge unique
1420 identifier, extension, and context in a JSON blob as the extra string
1421 instead of the transferee channel name as the peer.
1423 * ATTENDEDTRANSFER events now report the peer as NULL and additional
1424 information in the 'extra' string as a JSON blob. For transfers that occur
1425 between two bridged channels, the 'extra' JSON blob contains the primary
1426 bridge unique identifier, the secondary channel name, and the secondary
1427 bridge unique identifier. For transfers that occur between a bridged channel
1428 and a channel running an app, the 'extra' JSON blob contains the primary
1429 bridge unique identifier, the secondary channel name, and the app name.
1431 * LOCAL_OPTIMIZE events have been added to convey local channel
1432 optimizations with the record occurring for the semi-one channel and
1433 the semi-two channel name in the peer field.
1435 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
1436 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
1437 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
1438 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
1439 regardless of whether or not that bridge happens to contain multiple
1444 * When compiled with '--enable-dev-mode', the astobj2 library will now add
1445 several CLI commands that allow for inspection of ao2 containers that
1446 register themselves with astobj2. The CLI commands are 'astobj2 container
1447 dump', 'astobj2 container stats', and 'astobj2 container check'.
1449 * Added specific CLI commands for bridge inspection. This includes 'bridge
1450 show all', which lists all bridges in the system, and 'bridge show {id}',
1451 which provides specific information about a bridge.
1453 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
1454 ejecting the channels currently in the bridge. If the channels cannot
1455 continue in the dialplan or application that put them in the bridge, they
1458 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1460 * Added commands to inspect and manipulate the registered bridge technologies.
1461 This include 'bridge technology show', which lists the registered bridge
1462 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1463 which controls whether or not a registered bridge technology can be used
1464 during smart bridge operations. If a technology is suspended, it will not
1465 be used when a bridge technology is picked for channels; when unsuspended,
1466 it can be used again.
1468 * The command 'config show help {module} {type} {option}' will show
1469 configuration documentation for modules with XML configuration
1470 documentation. When {module}, {type}, and {option} are omitted, a listing
1471 of all modules with registered documentation is displayed. When {module}
1472 is specified, a listing of all configuration types for that module is
1473 displayed, along with their synopsis. When {module} and {type} are
1474 specified, a listing of all configuration options for that type are
1475 displayed along with their synopsis. When {module}, {type}, and {option}
1476 are specified, detailed information for that configuration option is
1479 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1480 a listing of all installed media sounds available on the system and
1481 detailed information about a sound, respectively.
1483 * 'xmldoc dump' has been added. This CLI command will dump the XML
1484 documentation DOM as a string to the specified file. The Asterisk core
1485 will populate certain XML elements pulled from the source files with
1486 additional run-time information; this command lets a user produce the
1487 XML documentation with all information.
1491 * Parking has been pulled from core and placed into a separate module called
1492 res_parking. See Parking changes below for more details. Configuration for
1493 parking should now be performed in res_parking.conf. Configuration for
1494 parking in features.conf is now unsupported.
1496 * Core attended transfers now have several new options. While performing an
1497 attended transfer, the transferer now has the following options:
1498 - *1 - cancel the attended transfer (configurable via atxferabort)
1499 - *2 - complete the attended transfer, dropping out of the call
1500 (configurable via atxfercomplete)
1501 - *3 - complete the attended transfer, but stay in the call. This will turn
1502 the call into a multi-party bridge (configurable via atxferthreeway)
1503 - *4 - swap to the other party. Once an attended transfer has begun, this
1504 options may be used multiple times (configurable via atxferswap)
1506 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1507 must be on the channel initiating the transfer to have any effect.
1509 * The BRIDGE_FEATURES channel variable would previously only set features for
1510 the calling party and would set this feature regardless of whether the
1511 feature was in caps or in lowercase. Use of a caps feature for a letter
1512 will now apply the feature to the calling party while use of a lowercase
1513 letter will apply that feature to the called party.
1515 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1517 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1518 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1519 activated the dynamic feature.
1521 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1522 only on the channel executing the dynamic feature. Executing a dynamic
1523 feature on the bridge peer in a multi-party bridge will execute it on all
1524 peers of the activating channel.
1526 * You can now have the settings for a channel updated using the FEATURE()
1527 and FEATUREMAP() functions inherited to child channels by setting
1528 FEATURE(inherit)=yes.
1530 * automixmon now supports additional channel variables from automon including:
1531 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1532 and TOUCH_MIXMONITOR_MESSAGE_STOP
1534 * A new general features.conf option 'recordingfailsound' has been added which
1535 allowssetting a failure sound for a user tries to invoke a recording feature
1536 such as automon or automixmon and it fails.
1538 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1539 features.c for atxferdropcall=no to work properly. This option now just
1544 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1545 be used. Given that this can cause the Asterisk log files to grow quickly,
1546 this option should only be used if an external mechanism for log management
1551 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1552 will store the path information for that peer when it registers. Realtime
1553 tables can also use the 'supportpath' field to enable Path header support.
1555 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1556 objectIdentifier. This maps to the supportpath option in sip.conf.
1560 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1561 provides modules a useful abstraction on top of the many storage mechanisms
1562 in Asterisk, including the Asterisk Database, static configuration files,
1563 static Realtime, and dynamic Realtime. It also provides a caching service.
1564 Users can configure a hierarchy of data storage layers for specific modules
1567 * All future modules which utilize Sorcery for object persistence must have a
1568 column named "id" within their schema when using the Sorcery realtime module.
1569 This column must be able to contain a string of up to 128 characters in length.
1571 Security Events Framework
1573 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1574 the "seconds-microseconds" format that it was using previously.
1578 * The Stasis message bus is a publish/subscribe message bus internal to
1579 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1580 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1581 Stasis can be configured in stasis.conf. Note that these parameters operate
1582 at a very low level in Asterisk, and generally will not require changes.
1586 * When a channel driver is configured to enable jiterbuffers, they are now
1587 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1588 is already set for that channel when it enters, such as by the JITTERBUFFER
1589 function, then the existing jitterbuffer will be used and the one set by
1590 the channel driver will not be applied.
1594 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1595 dialplan applications provided by the app_agent_pool module. Agents are
1596 connected with callers using the new AgentRequest dialplan application.
1597 The Agents:<agent-id> device state is available to monitor the status of an
1598 agent. See agents.conf.sample for valid configuration options.
1600 * The updatecdr option has been removed. Altering the names of channels on a
1601 CDR is not supported - the name of the channel is the name of the channel,
1602 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1603 has also been removed, for the same reason.
1605 * The endcall and enddtmf configuration options are removed. Use the
1606 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1607 channel before calling AgentLogin.
1611 * chan_bridge has been removed. Its functionality has been incorporated
1612 directly into the ConfBridge application itself.
1616 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1617 of the specified span and its B-channels. Note that this command should
1618 only be used if you understand the risks it entails.
1620 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1621 A range of channels can be specified to be destroyed. Note that this command
1622 should only be used if you understand the risks it entails.
1624 * Added the CLI command 'dahdi create channels'. A range of channels can be
1625 specified to be created, or the keyword 'new' can be used to add channels
1628 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1629 the exact configured mailbox name. For app_voicemail mailboxes this is
1632 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1636 * IPv6 support has been added. We are now able to bind to and
1637 communicate using IPv6 addresses.
1641 * The /b option has been removed.
1643 * chan_local moved into the system core and is no longer a loadable module.
1647 * Added general support for busy detection.
1649 * Added ECAM command support for Sony Ericsson phones.
1653 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1654 SIP stack. A collection of resource modules provides the bulk of the SIP
1655 functionality. For more information on the new SIP channel driver, see
1656 https://wiki.asterisk.org/wiki/x/JYGLAQ
1660 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1661 using the 'supportpath' setting, either on a global basis or on a peer basis.
1662 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1663 set of proxies by using a pre-loaded route-set defined by the Path headers in
1664 the REGISTER request. See Realtime updates for more configuration information.
1666 * The SIP_CODEC family of variables may now specify more than one codec. Each
1667 codec must be separated by a comma. The first codec specified is the
1668 preferred codec for the offer. This allows a dialplan writer to specify both
1669 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1671 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1672 in the core, and can be filtered out using the 'eventfilter' parameter
1675 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1676 codecs configured for a peer instead of the requested codec.
1678 * The option "register_retry_403" has been added to chan_sip to work around
1679 servers that are known to erroneously send 403 in response to valid
1680 REGISTER requests and allows Asterisk to continue attepmting to connect.
1684 * Added the 'immeddialkey' parameter. If set, when the user presses the
1685 configured key the already entered number will be immediately dialed. This
1686 is useful when the dialplan allows for variable length pattern matching.
1687 Valid options are '*' and '#'.
1689 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1690 milliseconds) before a call forward is considered to not be answered.
1692 * The 'serviceurl' parameter allows Service URLs to be attached to line
1701 * The password option has been disabled, as the AgentLogin application no
1702 longer provides authentication.
1706 * Due to changes in the Asterisk core, this function is no longer needed to
1707 preserve a MixMonitor on a channel during transfer operations and dialplan
1708 execution. It is effectively obsolete.
1712 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1713 deprecated. Use the CHANNEL function instead to access these attributes.
1715 * The 'l' option has been removed. When reading a CDR attribute, the most
1716 recent record is always used. When writing a CDR attribute, all non-finalized
1719 * The 'r' option has been removed, for the same reason as the 'l' option.
1721 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1726 * A new function CDR_PROP has been added. This function lets you set properties
1727 on a channel's active CDRs. This function is write-only. Properties accept
1728 boolean values to set/clear them on the channel's CDRs. Valid properties
1730 - 'party_a' - make this channel the preferred Party A in any CDR between two
1731 channels. If two channels have this property set, the creation time of the
1732 channel is used to determine who is Party A. Note that dialed channels are
1733 never Party A in a CDR.
1734 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1735 application when set to True, and analogous to the 'e' option in ResetCDR
1740 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1741 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1742 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1745 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1746 string, i.e., [[context],extension],priority. If set on a channel, if a
1747 channel leaves a bridge but is not hung up it will resume dialplan execution
1752 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1753 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1754 The value of this setting is ignored when disabled is used for the argument.
1758 * A new function provided by chan_pjsip, this function can be used in
1759 conjunction with the Dial application to construct a dial string that will
1760 dial all contacts on an Address of Record associated with a chan_pjsip
1765 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1766 outbound channel prior to dialing.
1770 * Redirecting reasons can now be set to arbitrary strings. This means
1771 that the REDIRECTING dialplan function can be used to set the redirecting
1772 reason to any string. It also allows for custom strings to be read as the
1773 redirecting reason from SIP Diversion headers.
1777 * The SPEECH_ENGINE function now supports read operations. When read from, it
1778 will return the current value of the requested attribute.
1782 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1783 system as mailbox@context. The rest of the system cannot add @default
1784 to mailbox identifiers for app_voicemail that do not specify a context
1785 any longer. It is a mailbox identifier format that should only be
1786 interpreted by app_voicemail.
1792 res_agi (Asterisk Gateway Interface)
1794 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1796 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1799 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1800 will start the playback of the audio at the position specified. It will
1801 also return the final position of the file in 'endpos'.
1803 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1804 channel variable if the user stopped the file playback or if a remote
1805 entity stopped the playback. If neither stopped the playback, it will
1806 indicate the overall success/failure of the playback. If stopped early,
1807 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1810 * The SAY ALPHA command now accepts an additional parameter to control
1811 whether it specifies the case of uppercase, lowercase, or all letters to
1812 provide functionality similar to SayAlphaCase.
1814 res_ari (Asterisk RESTful Interface) (and others)
1816 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1817 control telephony primitives in Asterisk by remote client. This includes
1818 channels, bridges, endpoints, media, and other fundamental concepts. Users
1819 of ARI can develop their own communications applications, controlling
1820 multiple channels using an HTTP RESTful interface and receiving JSON events
1821 about the objects via a WebSocket connection. ARI can be configured in
1822 Asterisk via ari.conf. For more information on ARI, see
1823 https://wiki.asterisk.org/wiki/x/0YCLAQ
1827 * Parking has been extracted from the Asterisk core as a loadable module,
1828 res_parking. Configuration for parking is now provided by res_parking.conf.
1829 Configuration through features.conf is no longer supported.
1831 * res_parking uses the configuration framework. If an invalid configuration is
1832 supplied, res_parking will fail to load or fail to reload. Previously,
1833 invalid configurations would generally be accepted, with certain errors
1834 resulting in individually disabled parking lots.
1836 * Parked calls are now placed in bridges. While this is largely an
1837 architectural change, it does have implications on how channels in a parking
1838 lot are viewed. For example, commands that display channels in bridges will
1839 now also display the channels in a parking lot.
1841 * The order of arguments for the new parking applications have been modified.
1842 Timeout and return context/exten/priority are now implemented as options,
1843 while the name of the parking lot is now the first parameter. See the
1844 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1845 in-depth information as well as syntax.
1847 * Extensions are by default no longer automatically created in the dialplan to
1848 park calls or pickup parked calls. Generation of dialplan extensions can be
1849 enabled using the 'parkext' configuration option.
1851 * ADSI functionality for parking is no longer supported. The 'adsipark'
1852 configuration option has been removed as a result.
1854 * The PARKINGSLOT channel variable has been deprecated in favor of
1855 PARKING_SPACE to match the naming scheme of the new system.
1857 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1858 channel even when the configuration option 'comebactoorigin' is enabled.
1860 * A new CLI command 'parking show' has been added. This allows a user to
1861 inspect the parking lots that are currently in use.
1862 'parking show <parkinglot>' will also show the parked calls in a specific
1865 * The CLI command 'parkedcalls' is now deprecated in favor of
1866 'parking show <parkinglot>'.
1868 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1869 can be used to get a list of parked calls for a specific parking lot.
1871 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1872 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1873 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1874 longer a required argument.
1876 * The ParkAndAnnounce application is now provided through res_parking instead
1877 of through the separate app_parkandannounce module.
1879 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1880 by default. Instead, it will follow the timeout rules of the parking lot. The
1881 old behavior can be reproduced by using the 'c' option.
1883 * Dynamic parking lots will now fail to be created under the following
1885 - if the parking lot specified by PARKINGDYNAMIC does not exist
1886 - if they require exclusive park and parkedcall extensions which overlap
1887 with existing parking lots.
1889 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1890 currently contain no calls. Dynamic parking lots containing parked calls
1891 will persist through the reloads without alteration.
1893 * If 'parkext_exclusive' is set for a parking lot and that extension is
1894 already in use when that parking lot tries to register it, this is now
1895 considered a parking system configuration error. Configurations which do
1896 this will be rejected.
1898 * Added channel variable PARKER_FLAT. This contains the name of the extension
1899 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1900 comebacktoorigin is disabled, but the dialplan or an external control
1901 mechanism wants to use the extension in the park-dial context that was
1902 generated to re-dial the parker on timeout.
1904 res_pjsip (and many others)
1906 * A large number of resource modules make up the SIP stack based on pjsip.
1907 The chan_pjsip channel driver users these resource modules to provide
1908 various SIP functionality in Asterisk. The majority of configuration for
1909 these modules is performed in pjsip.conf. Other modules may use their
1910 own configuration files.
1912 * Added 'set_var' option for an endpoint. For each variable specified that
1913 variable gets set upon creation of a channel involving the endpoint.
1917 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1918 them, an Asterisk-specific version of PJSIP needs to be installed.
1919 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1921 res_statsd/res_chan_stats
1923 * A new resource module, res_statsd, has been added, which acts as a statsd
1924 client. This module allows Asterisk to publish statistics to a statsd
1925 server. In conjunction with res_chan_stats, it will publish statistics about
1926 channels to the statsd server. It can be configured via res_statsd.conf.
1930 * Device state for XMPP buddies is now available using the following format:
1931 XMPP/<client name>/<buddy address>
1932 If any resource is available the device state is considered to be not in use.
1933 If no resources exist or all are unavailable the device state is considered
1940 Realtime/Database Scripts
1942 * Asterisk previously included example db schemas in the contrib/realtime/
1943 directory of the source tree. This has been replaced by a set of database
1944 migrations using the Alembic framework. This allows you to use alembic to
1945 initialize the database for you. It will also serve as a database migration
1946 tool when upgrading Asterisk in the future.
1948 See contrib/ast-db-manage/README.md for more details.
1952 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1953 This python script will convert an existing sip.conf file to a
1954 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1955 is meant to be an aid in converting an existing chan_sip configuration to
1956 a chan_pjsip configuration, but it is expected that configuration beyond
1957 what the script provides will be needed.
1959 ------------------------------------------------------------------------------
1960 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1961 ------------------------------------------------------------------------------
1965 * The Asterisk build system will now build and install a shared library
1966 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1967 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1968 that Asterisk can ensure that these functions do *not* get called by any
1969 modules that are loaded into Asterisk, since they should only be called once
1970 in any single process. If desired, this feature can be disabled by supplying
1971 the "--disable-asteriskssl" option to the configure script.
1973 * A new make target, 'full', has been added to the Makefile. This performs
1974 the same compilation actions as make all, but will also scan the entirety of
1975 each source file for documentation. This option is needed to generate AMI
1976 event documentation. Note that your system must have Python in order for
1977 this make target to succeed.
1979 * The optimization portion of the build system has been reworked to avoid
1980 broken builds on certain architectures. All architecture-specific
1981 optimization has been removed in favor of using -march=native to allow gcc
1982 to detect the environment in which it is running when possible. This can
1983 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1985 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1986 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1988 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1989 previously parsed the header file to obtain the version of Asterisk, you
1990 will now have to go through Asterisk to get the version information.
1998 * Added 'F()' option. Similar to the dial option, this can be supplied with
1999 arguments indicating where the callee should go after the caller is hung up,
2000 or without options specified, the priority after the Queue will be used.
2005 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2006 all non-admin participants on a conference. The confbridge configuration
2007 file also allows for the default sounds played to all conference users when
2008 this occurs to be overriden using sound_participants_unmuted and
2009 sound_participants_muted.
2011 * Added menu action participant_count. This will playback the number of
2012 current participants in a conference.
2014 * Added announcement configuration option to user profile. If set the sound
2015 file will be played to the user, and only the user, upon joining the
2018 * Added record_file_append option that defaults to "yes", but if set to no
2019 will create a new file between each start/stop recording.
2024 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2025 channels respectively before the callee channels are called.
2030 * Added support for IPv6.
2032 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2033 external process will cause the current playlist to be cleared, including
2034 stopping any audio file that is currently playing. This is useful when you
2035 want to interrupt audio playback only when specific DTMF is entered by the
2041 * A new option, 'I' has been added to app_followme. By setting this option,
2042 Asterisk will not update the caller with connected line changes when they
2043 occur. This is similar to app_dial and app_queue.
2045 * The 'N' option is now ignored if the call is already answered.
2047 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2048 and caller channels respectively before the callee channels are called.
2050 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2056 * MixMonitor hooks now have IDs associated with them which can be used to
2057 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2058 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2059 now accepts that ID as an argument.
2061 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2062 indicated mailboxes.
2067 * The connect action in app_mysql now allows you to specify a port number to
2068 connect to. This is useful if you run a MySQL server on a non-standard
2074 * Increased the default number of allowed destinations from 5 to 12.
2079 * The app_page application now no longer depends on DAHDI or app_meetme. It
2080 has been re-architected to use app_confbridge internally.
2085 * Added queue options autopausebusy and autopauseunavail for automatically
2086 pausing a queue member when their device reports busy or congestion.
2088 * The 'ignorebusy' option for queue members has been deprecated in favor of
2089 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2090 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2091 per interface basis. Individual ringinuse values can now be set in
2092 queues.conf via an argument to member definitions. Lastly, the queue
2093 'ringinuse' setting now only determines defaults for the per member
2094 'ringinuse' setting and does not override per member settings like it does
2095 in earlier versions.
2097 * Added 'F()' option. Similar to the dial option, this can be supplied with
2098 arguments indicating where the callee should go after the caller is hung up,
2099 or without options specified, the priority after the Queue will be used.
2101 * Added new option log_member_name_as_agent, which will cause the membername to
2102 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2103 state_interface has been set.
2105 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2107 * App_queue will now play periodic announcements for the caller that
2108 holds the first position in the queue while waiting for answer.
2112 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2113 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2114 changed arguments to SayUnixTime so that every option is truly optional even
2115 when using multiple options (so that j option could be used without having to
2116 manually specify timezone and format) There are other benefits, e.g., format
2117 can now be used without specifying time zone as well.
2122 * Addition of the VM_INFO function - see Function changes.
2124 * The imapserver, imapport, and imapflags configuration options can now be
2125 overriden on a user by user basis.
2127 * When voicemail plays a message's envelope with saycid set to yes, when
2128 reaching the caller id field it will play a recording of a file with the same
2129 base name as the sender's callerid if there is a similarly named file in
2130 <astspooldir>/recordings/callerids/
2132 * Voicemails now contains a unique message identifier "msg_id", which is stored
2133 in the message envelope with the sound files. IMAP backends will now store
2134 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2135 backends will store the message identifier in a "msg_id" column. See
2136 UPGRADE.txt for more information.
2138 * Added VoiceMailPlayMsg application. This application will play a single
2139 voicemail message from a mailbox. The result of the application, SUCCESS or
2140 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2145 * Hangup handlers can be attached to channels using the CHANNEL() function.
2146 Hangup handlers will run when the channel is hung up similar to the h
2147 extension. The hangup_handler_push option will push a GoSub compatible
2148 location in the dialplan onto the channel's hangup handler stack. The
2149 hangup_handler_pop option will remove the last added location, and optionally
2150 replace it with a new GoSub compatible location. The hangup_handler_wipe
2151 option will remove all locations on the stack, and optionally add a new
2154 * The expression parser now recognizes the ABS() absolute value function,
2155 which will convert negative floating point values to positive values.
2157 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2158 control of faxdetect.
2160 * Addition of the VM_INFO function that can be used to retrieve voicemail
2161 user information, such as the email address and full name.
2162 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2165 * The REDIRECTING function now supports the redirecting original party id
2168 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2169 lets you set some of the configuration options from the [general] section
2170 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2171 the key sequence used to activate built-in features, such as blindxfer,
2172 and automon. See the built-in documentation for details.
2174 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2175 instead of simply the uri. This is the format that MessageSend() can use
2176 in the from parameter for outgoing SIP messages.
2178 * Added the PRESENCE_STATE function. This allows retrieving presence state
2179 information from any presence state provider. It also allows setting
2180 presence state information from a CustomPresence presence state provider.
2181 See AMI/CLI changes for related commands.
2183 * Added the AMI_CLIENT function to make manager account attributes available
2184 to the dialplan. It currently supports returning the current number of
2185 active sessions for a given account.
2187 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2188 and the REDIRECTING functions.
2196 * Added a manager event "LocalBridge" for local channel call bridges between
2197 the two pseudo-channels created.
2202 * Added dialtone_detect option for analog ports to disconnect incoming
2203 calls when dialtone is detected.
2205 * Added option colp_send to send ISDN connected line information. Allowed
2206 settings are block, to not send any connected line information; connect, to
2207 send connected line information on initial connect; and update, to send
2208 information on any update during a call. Default is update.
2210 * Add options namedcallgroup and namedpickupgroup to support installations
2211 where a higher number of groups (>64) is required.
2213 * Added support to use private party ID information with PRI calls.
2218 * A new channel driver named chan_motif has been added which provides support for
2219 Google Talk and Jingle in a single channel driver. This new channel driver includes
2220 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2221 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2222 specification, current Google Jingle specification, and the original Google Talk
2228 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2229 globally and overriden on a peer by peer basis.
2231 * Direct media functionality has been added. Options in config are:
2232 directmedia (directrtp) and directrtpsetup (earlydirect)
2234 * ChannelUpdate events now contain a CallRef header.
2239 * Asterisk will no longer substitute CID number for CID name in the display
2240 name field if CID number exists without a CID name. This change improves
2241 compatibility with certain device features such as Avaya IP500's directory
2244 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2245 created using that setting to not be removed during SIP reload.
2247 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2248 request with a "Record:" header, this will turn the requested feature on/off.
2249 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2250 dynamic features must be enabled and configured properly on the requesting
2251 channel for this to function properly.
2253 * Add support to realtime for the 'callbackextension' option.
2255 * When multiple peers exist with the same address, but differing
2256 callbackextension options, incoming requests that are matched by address
2257 will be matched to the peer with the matching callbackextension if it is
2260 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2261 which set the force_rport and comedia options automatically if Asterisk
2262 detects that an incoming SIP request crossed a NAT after being sent by
2263 the remote endpoint.
2265 * The default global nat setting in sip.conf has been changed from force_rport
2266 to auto_force_rport.
2268 * NAT settings are now a combinable list of options. The equivalent of the
2269 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2271 * Adds an option send_diversion which can be disabled to prevent
2272 diversion headers from automatically being added to INVITE requests.
2274 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2275 be sent to the remote host at a given interval to keep the NAT mapping open.
2276 This can be enabled using the keepalive configuration option.
2278 * Add option 'tonezone' to specify country code for indications. This option
2279 can be set both globally and overridden for specific peers.
2281 * The SIP Security Events Framework now supports IPv6.
2283 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2284 between multiple user agents. When set, for directmedia reinvites,
2285 Asterisk will not send an immediate reinvite on an incoming call leg. This
2286 option is useful when peered with another SIP user agent that is known to
2287 send immediate direct media reinvites upon call establishment.
2289 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2292 * Add options subminexpiry and submaxexpiry to set limits of subscription
2293 timer independently from registration timer settings. The setting of the
2294 registration timer limits still is done by options minexpiry, maxexpiry
2295 and defaultexpiry. For backwards compatibility the setting of minexpiry
2296 and maxexpiry also is used to configure the subscription timer limits if
2297 subminexpiry and submaxexpiry are not set in sip.conf.
2299 * Set registration timer limits to default values when reloading sip
2300 configuration and values are not set by configuration.
2302 * Add options namedcallgroup and namedpickupgroup to support installations
2303 where a higher number of groups (>64) is required.
2305 * When a MESSAGE request is received, the address the request was received from
2306 is now saved in the SIP_RECVADDR variable.
2308 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2309 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2310 the ANI2/OLI information is set on the channel, which can be retrieved using
2311 the CALLERID function.
2313 * Peers can now be configured to support negotiation of ICE candidates using
2314 the setting icesupport. See res_rtp_asterisk changes for more information.
2316 * Added support for format attribute negotiation. See the Codecs changes for
2319 * Extra headers specified with SIPAddHeader are sent with the REFER message
2320 when using Transfer application. See refer_addheaders in sip.conf.sample.
2322 * Added support to use private party ID information with calls.
2324 * Adds an option discard_remote_hold_retrieval that when set stops telling
2325 the peer to start music on hold.
2330 * Added skinny version 17 protocol support.
2334 --------------------
2335 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2337 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2338 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2339 as per the UNISTIM protocol.
2341 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2342 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2344 * Added ability to use multiple lines for a single phone. This allows multiple
2345 calls to occur on a single phone, using callwaiting and switching between calls.
2347 * Added option 'sharpdial' allowing end dialing by pressing # key
2349 * Added option 'interdigit_timer' to control phone dial timeout
2351 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2353 * Added global 'debug' option, that enables debug in channel driver
2355 * Added ability to translate on-screen menu in multiple languages. Tested on
2356 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2357 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2360 * In addition to English added French and Russian languages for on-screen menus
2362 * Reworked dialing number input: added dialing by timeout, immediate dial on
2363 on dialplan compare, phone number length now not limited by screen size
2365 * Added ability to pickup a call using features.conf defined value and
2371 * Add options namedcallgroup and namedpickupgroup to support installations
2372 where a higher number of groups (>64) is required.
2374 * Added support to use private party ID information with calls.
2379 * The minimum DTMF duration can now be configured in asterisk.conf
2380 as "mindtmfduration". The default value is (as before) set to 80 ms.
2381 (previously it was only available in source code)
2383 * Named ACLs can now be specified in acl.conf and used in configurations that
2384 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2385 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2386 working ACL. In addition, some CLI commands have been added to provide
2387 show information and allow for module reloading - see CLI Changes.
2389 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2390 items (separated by commas), and items in the rule can be negated by prefixing
2391 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2392 longer necessray to control the order that the 'permit' and 'deny' columns are
2393 returned from queries.
2395 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2396 be used within the dynamic weight attribute when specifying a mapping.
2398 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2399 header, instead of putting the user defined event name there. When enabled
2400 the UserDefType header is added for user defined events. This feature is
2401 enabled with the setting show_user_defined.
2403 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2404 line purposes use the following variables instead of their macro equivalents:
2405 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2406 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
2407 cc_callback_macro in channel configurations.
2409 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
2412 * Call files now support the "early_media" option to connect with an outgoing
2413 extension when early media is received.
2415 * Added support to use private party ID information with calls.
2420 * A new channel variable, AGIEXITONHANGUP, has been added which allows
2421 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
2422 AGI application would exit immediately after a channel hangup is detected.
2424 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
2425 are resolved and each address is attempted in turn until one succeeds or
2429 AMI (Asterisk Manager Interface)
2431 * The originate action now has an option "EarlyMedia" that enables the
2432 call to bridge when we get early media in the call. Previously,
2433 early media was disregarded always when originating calls using AMI.
2435 * Added setvar= option to manager accounts (much like sip.conf)
2437 * Originate now generates an error response if the extension given is not found
2440 * MixMonitor will now show IDs associated with the mixmonitor upon creating
2441 them if the i(variable) option is used. StopMixMonitor will accept
2442 MixMonitorID as an option to close specific MixMonitors.
2444 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
2445 updated to include information about peers configured with
2446 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
2447 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
2448 returned if auto_force_rport is not enabled.
2450 * Added SIPpeerstatus manager command which will generate PeerStatus events
2451 similar to the existing PeerStatus events found in chan_sip on demand.
2453 * Hangup now can take a regular expression as the Channel option. If you want
2454 to hangup multiple channels, use /regex/ as the Channel option. Existing
2455 behavior to hanging up a single channel is unchanged, but if you pass a regex,
2456 the manager will send you a list of channels back that were hung up.
2458 * Support for IPv6 addresses has been added.
2460 * AMI Events can now be documented in the Asterisk source. Note that AMI event
2461 documentation is only generated when Asterisk is compiled using 'make full'.
2462 See the CLI section for commands to display AMI event information.
2464 * The AMI Hangup event now includes the AccountCode header so you can easily
2465 correlate with AMI Newchannel events.
2467 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2468 the StateInterface of the queue member.
2470 * Added AMI event SessionTimeout in the Call category that is issued when a
2471 call is terminated due to either RTP stream inactivity or SIP session timer
2474 * CEL events can now contain a user defined header UserDefType. See core
2475 changes for more information.
2477 * OOH323 ChannelUpdate events now contain a CallRef header.
2479 * Added PresenceState command. This command will report the presence state for
2480 the given presence provider.
2482 * Added Parkinglots command. This will list all parking lots as a series of
2483 AMI Parkinglot events.
2485 * Added MessageSend command. This behaves in the same manner as the
2486 MessageSend application, and is a technolgoy agnostic mechanism to send out
2487 of call text messages.
2489 * Added "message" class authorization. This grants an account permission to
2490 send out of call messages. Write-only.
2495 * The "dialplan add include" command has been modified to create context a context
2496 if one does not already exist. For instance, "dialplan add include foo into bar"
2497 will create context "bar" if it does not already exist.
2499 * A "dialplan remove context" command has been added to remove a context from
2502 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2503 filenames of all running mixmonitors on a channel.
2505 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2506 numeric instead of 0, 1, or 2.
2508 * "stun show status" will show a table describing how the STUN client is
2511 * "acl show [named acl]" will show information regarding a Named ACL. The
2512 acl module can be reloaded with "reload acl".
2514 * Added CLI command to display AMI event information - "manager show events",
2515 which shows a list of all known and documented AMI events, and "manager show
2516 event [event name]", which shows detail information about a specific AMI
2519 * The result of the CLI command "queue show" now includes the state interface
2520 information of the queue member.
2522 * The command "core set verbose" will now set a separate level of logging for
2523 each remote console without affecting any other console.
2525 * Added command "cdr show pgsql status" to check connection status
2527 * "sip show channel" will now display the complete route set.
2529 * Added "presencestate list" command. This command will list all custom
2530 presence states that have been set by using the PRESENCE_STATE dialplan
2533 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2534 command. This changes a custom presence to a new state.
2539 * Codec lists may now be modified by the '!' character, to allow succinct
2540 specification of a list of codecs allowed and disallowed, without the
2541 requirement to use two different keywords. For example, to specify all
2542 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2544 * Add support for parsing SDP attributes, generating SDP attributes, and
2545 passing it through. This support includes codecs such as H.263, H.264, SILK,
2546 and CELT. You are able to set up a call and have attribute information pass.
2547 This should help considerably with video calls.
2549 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2550 just like the GSM codec.
2554 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2555 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2559 * Asterisk version and build information is now logged at the beginning of a
2562 * Threads belonging to a particular call are now linked with callids which get
2563 added to any log messages produced by those threads. Log messages can now be
2564 easily identified as involved with a certain call by looking at their call id.
2565 Call ids may also be attached to log messages for just about any case where
2566 it can be determined to be related to a particular call.
2568 * Each logging destination and console now have an independent notion of the
2569 current verbosity level. Logger.conf now allows an optional argument to
2570 the 'verbose' specifier, indicating the level of verbosity sent to that
2571 particular logging destination. Additionally, remote consoles now each
2572 have their own verbosity level. The command 'core set verbose' will now set
2573 a separate level for each remote console without affecting any other
2579 * Added 'announcement' option which will play at the start of MOH and between
2580 songs in modes of MOH that can detect transitions between songs (eg.
2586 * New per parking lot options: comebackcontext and comebackdialtime. See
2587 configs/features.conf.sample for more details.
2589 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2592 * Channel variable PARKEDCALL is now set with the name of the parking lot
2593 when a timeout occurs.
2599 CDR Postgresql Driver
2601 * Added command "cdr show pgsql status" to check connection status
2604 CDR Adaptive ODBC Driver
2606 * Added schema option for databases that support specifying a schema.
2614 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2615 CALENDAR_WRITE has completed successfully.
2620 * A new option, 'probation' has been added to rtp.conf
2621 RTP in strictrtp mode can now require more than 1 packet to exit learning
2622 mode with a new source (and by default requires 4). The probation option
2623 allows the user to change the required number of packets in sequence to any
2624 desired value. Use a value of 1 to essentially restore the old behavior.
2625 Also, with strictrtp on, Asterisk will now drop all packets until learning
2626 mode has successfully exited. These changes are based on how pjmedia handles
2627 media sources and source changes.
2629 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2630 enabled or disabled using the icesupport setting. A variety of other
2631 settings have been introduced to configure STUN/TURN connections.
2636 * A new module, res_corosync, has been introduced. This module uses the
2637 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2638 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2639 Device State (presence) information. This module is very similar to, and
2640 is a replacement for the res_ais module that was in previous releases of
2646 * This module adds a cleaned up, drop-in replacement for res_jabber called
2647 res_xmpp. This provides the same externally facing functionality but is
2648 implemented differently internally. res_jabber has been deprecated in favor
2649 of res_xmpp; please see the UPGRADE.txt file for more information.
2654 * The safe_asterisk script has been updated to allow several of its parameters
2655 to be set from environment variables. This also enables a custom run
2656 directory of Asterisk to be specified, instead of defaulting to /tmp.
2658 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2659 its value to determine the directory to assume is the top-level directory of
2660 the source tree. If the variable is not set, it defaults to the current
2661 behavior and uses the current working directory.
2663 ------------------------------------------------------------------------------
2664 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2665 ------------------------------------------------------------------------------
2669 * Asterisk now has protocol independent support for processing text messages
2670 outside of a call. Messages are routed through the Asterisk dialplan.
2671 SIP MESSAGE and XMPP are currently supported. There are options in
2672 jabber.conf and sip.conf to allow enabling these features.
2673 -> jabber.conf: see the "sendtodialplan" and "context" options.
2674 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2675 and "outofcall_message_context" options.
2676 The MESSAGE() dialplan function and MessageSend() application have been
2677 added to go along with this functionality. More detailed usage information
2678 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2679 * If real-time text support (T.140) is negotiated, it will be preferred for
2680 sending text via the SendText application. For example, via SIP, messages
2681 that were once sent via the SIP MESSAGE request would be sent via RTP if
2682 T.140 text is negotiated for a call.
2686 * parkedmusicclass can now be set for non-default parking lots.
2688 Asterisk Manager Interface
2689 --------------------------
2690 * PeerStatus now includes Address and Port.
2691 * Added Hold events for when the remote party puts the call on and off hold
2692 for chan_dahdi ISDN channels.
2693 * Added new action MeetmeListRooms to list active conferences (shows same
2694 data as "meetme list" at the CLI).
2695 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2696 Description field that is set by 'description' in the channel configuration
2698 * Added Uniqueid header to UserEvent.
2699 * Added new action FilterAdd to control event filters for the current session.
2700 This requires the system permission and uses the same filter syntax as
2701 filters that can be defined in manager.conf
2702 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2703 versions had some instances of the event converted, but others were left
2704 as-is. All Unlink events should now be converted to Bridge events. The AMI
2705 protocol version number was incremented to 1.2 as a result of this change.
2707 Asterisk HTTP Server
2708 --------------------------
2709 * The HTTP Server can bind to IPv6 addresses.
2712 --------------------------
2713 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2714 with busydetect. usage example: busypattern=200,200,200,600
2717 --------------------------
2718 * New 'gtalk show settings' command showing the current settings loaded from
2720 * The 'logger reload' command now supports an optional argument, specifying an
2721 alternate configuration file to use.
2722 * 'dialplan add extension' command will now automatically create a context if
2723 the specified context does not exist with a message indicated it did so.
2724 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2725 Description field which can be populated with 'description' in the channel
2726 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2729 --------------------------
2730 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2731 thus allowing records which do NOT match the specified filter.
2732 * Added ability to log CONGESTION calls to CDR
2735 --------------------------
2736 * Ability to define custom SILK formats in codecs.conf.
2737 * Addition of speex32 audio format with translation.
2738 * CELT codec pass-through support and ability to define
2739 custom CELT formats in codecs.conf.
2740 * Ability to read raw signed linear files with sample rates
2741 ranging from 8khz - 192khz. The new file extensions introduced
2742 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2743 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2744 Skinny, H.323, etc) can still only support the following codecs:
2745 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2746 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2747 Video: h261, h263, h263p, h264, mpeg4
2752 --------------------------
2753 * New highly optimized and customizable ConfBridge application capable of
2754 mixing audio at sample rates ranging from 8khz-96khz.
2755 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2756 and bridge profiles on a channel.
2757 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2758 about a conference such as locked status and number of parties, admins,
2760 * Addition of video_mode option in confbridge.conf for adding video support
2761 into a bridge profile.
2762 * Addition of the follow_talker video_mode in confbridge.conf. This video
2763 mode dynamically switches the video feed to always display the loudest talker
2764 supplying video in the conference.
2768 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2769 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2770 variables from asterisk.conf.
2774 * Addition of the JITTERBUFFER dialplan function. This function allows
2775 for jitterbuffering to occur on the read side of a channel. By using
2776 this function conference applications such as ConfBridge and MeetMe can
2777 have the rx streams jitterbuffered before conference mixing occurs.
2778 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2780 * Added STRREPLACE function. This function let's the user search a variable
2781 for a given string to replace with another string as many times as the
2782 user specifies or just throughout the whole string.
2783 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2784 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2785 * Added extensions to chan_ooh323 in function CHANNEL()
2787 libpri channel driver (chan_dahdi) DAHDI changes
2788 --------------------------
2789 * Added moh_signaling option to specify what to do when the channel's bridged
2790 peer puts the ISDN channel on hold.
2791 * Added display_send and display_receive options to control how the display ie
2792 is handled. To send display text from the dialplan use the SendText()
2793 application when the option is enabled.
2794 * Added mcid_send option to allow sending a MCID request on a span.
2797 --------------------------
2798 * Added setvar option to calendar.conf to allow setting channel variables on
2799 notification channels.
2800 * Added "calendar show types" CLI command to list registered calendar
2804 --------------------------
2805 * Added two new options, r and t with file name arguments to record
2806 single direction (unmixed) audio recording separate from the bidirectional
2807 (mixed) recording. The mixed file name argument is optional now as long
2808 as at least one recording option is used.
2811 --------------------------
2812 * Added a new option, l, which will disable local call optimization for
2813 channels involved with the FollowMe thread. Use this option to improve
2814 compatability for a FollowMe call with certain dialplan apps, options, and
2818 --------------------------
2819 * Added option "k" that will automatically close the conference when there's
2820 only one person left when a user exits the conference.
2823 --------------------------
2824 * cel_pgsql now supports the 'extra' column for data added using the
2825 CELGenUserEvent() application.
2828 --------------------------
2829 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2830 in the sample extensions.lua file for syntax details.
2831 * Applications that perform jumps in the dialplan such as Goto will now
2832 execute properly. When pbx_lua detects that the context, extension, or
2833 priority we are executing on has changed it will immediately return control
2834 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2835 the priority after the currently executing priority.
2836 * An autoservice is now started by default for pbx_lua channels. It can be
2837 stopped and restarted using the autoservice_stop() and autoservice_start()
2841 --------------------------
2842 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2843 into a FAXStatus event with an 'Operation' header that will be either
2844 'send', 'receive', and 'gateway'.
2845 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2846 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2847 feature will handle converting a fax call between an audio T.30 fax terminal
2848 and an IFP T.38 fax terminal.
2852 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2853 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2854 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2858 * Added general option negative_penalty_invalid default off. when set
2859 members are seen as invalid/logged out when there penalty is negative.
2860 for realtime members when set remove from queue will set penalty to -1.
2861 * Added queue option autopausedelay when autopause is enabled it will be
2862 delayed for this number of seconds since last successful call if there
2863 was no prior call the agent will be autopaused immediately.
2864 * Added member option ignorebusy this when set and ringinuse is not
2865 will allow per member control of multiple calls as ringinuse does for
2870 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2872 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2873 one participant left (much like a normal call bridge)
2874 * Added extra argument to Originate to set timeout.
2878 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2879 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2880 utility in the UTILS section of menuselect. If an existing astdb is found and no
2881 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2882 convert an existing astdb to the SQLite3 version automatically at runtime.
2886 * Modules marked as deprecated are no longer marked as building by default. Enabling
2887 these modules is still available via menuselect.
2891 * authdebug is now disabled by default. To enable this functionaility again
2892 set authdebug = yes in iax.conf.
2896 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2897 releases it was disabled.
2901 * The PBX core previously made a call with a non-existing extension test for
2902 extension s@default and jump there if the extension existed.
2903 This was a bad default behaviour and violated the principle of least surprise.
2904 It has therefore been changed in this release. It may affect some
2905 applications and configurations that rely on this behaviour. Most channel
2906 drivers have avoided this for many releases by testing whether the extension
2907 called exists before starting the PBX and generating a local error.
2908 This behaviour still exists and works as before.
2910 Extension "s" is used when no extension is given in a channel driver,
2911 like immediate answer in DAHDI or calling to a domain with no user part
2914 ------------------------------------------------------------------------------
2915 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2916 ------------------------------------------------------------------------------
2920 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2921 now defaults to force_rport. It is very important that phones requiring nat=no be
2922 specifically set as such instead of relying on the default setting. If at all
2923 possible, all devices should have nat settings configured in the general section as
2924 opposed to configuring nat per-device.
2925 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2926 codecs sent in response to an INVITE to the single most preferred codec.
2927 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2928 to be used for the outgoing call. It must be one of the codecs configured
2930 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2931 to be used for holding a private key. If tlsprivatekey is not specified,
2932 tlscertfile is searched for both public and private key.
2933 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2934 outbound client connections to be specified.
2935 * The sendrpid parameter has been expanded to include the options
2936 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2937 header to be sent (equivalent to setting sendrpid=yes) and setting
2938 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2939 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2940 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2941 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2942 will accept the SDP even if the SDP version number is not properly incremented,
2943 but will generate a warning in the log indicating that the SIP peer that sent
2944 the SDP should have the 'ignoresdpversion' option set.
2945 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2946 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2947 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2948 remote side requests it and disables symmetric RTP support. Setting it to
2949 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2950 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2951 and enables symmetric RTP support.
2952 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2953 response. This permits the master channel to know how each channel dialled
2954 in a multi-channel setup resolved in an individual way. This carries a
2955 performance penalty and can be disabled in sip.conf using the
2956 'storesipcause' option.
2957 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2958 configuration for the externip and externhost options when tcp or tls is used.
2959 * Added support for message body (stored in content variable) to SIP NOTIFY message
2960 accessible via AMI and CLI.
2961 * Added 'media_address' configuration option which can be used to explicitly specify
2962 the IP address to use in the SDP for media (audio, video, and text) streams.
2963 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2964 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2966 * Added 'use_q850_reason' configuration option for generating and parsing
2967 if available Reason: Q.850;cause=<cause code> header. It is implemented
2968 in some gateways for better passing PRI/SS7 cause codes via SIP.
2969 * When dialing SIP peers, a new component may be added to the end of the dialstring
2970 to indicate that a specific remote IP address or host should be used when dialing
2971 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2972 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2973 ability to selectively force bridged channels to also be encrypted is also
2974 implemented. Branching in the dialplan can be done based on whether or not
2975 a channel has secure media and/or signaling.
2976 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2978 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2979 Charge messages to snom phones.
2980 * Added support for G.719 media streams.
2981 * Added support for 16khz signed linear media streams.
2982 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2983 RTP has been outfitted with the same abilities.
2984 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2985 available in device configurations as well as in the dial plan.
2986 * Addition of the 'subscribe_network_change' option for turning on and off
2987 res_stun_monitor module support in chan_sip.
2988 * Addition of the 'auth_options_requests' option for turning on and off
2989 authentication for OPTIONS requests in chan_sip.
2993 * Add #tryinclude statement for config files. This provides the same
2994 functionality as the #include statement however an asterisk module will
2995 still load if the filename does not exist. Using the #include statement
2996 Asterisk will not allow the module to load.
3000 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3001 on realtime updates.
3002 * Added the ability for chan_iax2 to inform the dialplan whether or not
3003 encryption is being used. This interoperates with the SIP SRTP implementation
3004 so that a secure SIP call can be bridged to a secure IAX call when the
3005 dialplan requires bridged channels to be "secure".
3006 * Addition of the 'subscribe_network_change' option for turning on and off
3007 res_stun_monitor module support in chan_iax.
3012 * Added ability to preset channel variables on indicated lines with the setvar
3013 configuration option. Also, clearvars=all resets the list of variables back
3015 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3016 See configs/res_pktccops.conf for more information.
3018 XMPP Google Talk/Jingle changes
3019 -------------------------------
3020 * Added the externip option to gtalk.conf.
3021 * Added the stunaddr option to gtalk.conf which allows for the automatic
3022 retrieval of the external ip from a stun server.
3026 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3027 match to a partial channel name.
3028 * Added .m3u support for Mp3Player application.
3029 * Added progress option to the app_dial D() option. When progress DTMF is
3030 present, those values are sent immediately upon receiving a PROGRESS message
3031 regardless if the call has been answered or not.
3032 * Added functionality to the app_dial F() option to continue with execution
3033 at the current location when no parameters are provided.
3034 * Added the 'a' option to app_dial to answer the calling channel before any
3035 announcements or macros are executed.
3036 * Modified app_dial to set answertime when the called channel answers even if
3037 the called channel hangs up during playback of an announcement.
3038 * Modified app_dial 'r' option to support an additional parameter to play an
3039 indication tone from indications.conf
3040 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3041 to cycle through the next available channel. By default this is still '*'.
3042 * Added x() option to app_chanspy. This option allows DTMF to be set to
3043 exit the application.
3044 * The Voicemail application has been improved to automatically ignore messages
3045 that only contain silence.
3046 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3047 associated mailbox(es) to be greetings-only.
3048 * The ChanSpy application now has the 'S' option, which makes the application
3049 automatically exit once it hits a point where no more channels are available
3051 * The ChanSpy application also now has the 'E' option, which spies on a single
3052 channel and exits when that channel hangs up.
3053 * The MeetMe application now turns on the DENOISE() function by default, for
3054 each participant. In our tests, this has significantly decreased background
3055 noise (especially noisy data centers).
3056 * Voicemail now permits storage of secrets in a separate file, located in the
3057 spool directory of each individual user. The control for this is located in
3058 the "passwordlocation" option in voicemail.conf. Please see the sample
3059 configuration for more information.
3060 * The ChanIsAvail application now exposes the returned cause code using a separate
3061 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3062 * Added 'd' option to app_followme. This option disables the "Please hold"
3064 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3065 received will terminate recording.
3066 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3067 Previously the folder could only be set per context, but has now been extended
3068 using the imapfolder option.
3069 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3070 * Voicemail now allows the pager date format to be specified separately from the
3072 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3073 to allow joining, leaving, and sending text to group chats.
3074 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3075 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3076 to all paged phones (and optionally excluding the caller's one using the new
3077 option 'n') before the call is bridged.
3078 * The 'f' option to Dial has been augmented to take an optional argument. If no
3079 argument is provided, the 'f' option works as it always has. If an argument is
3080 provided, then the connected party information of all outgoing channels created
3081 during the Dial will be set to the argument passed to the 'f' option.
3082 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3084 * The OSP lookup application adds in/outbound network ID, optional security,
3085 number portability, QoS reporting, destination IP port, custom info and service
3087 * Added new application VMSayName that will play the recorded name of the voicemail
3088 user if it exists, otherwise will play the mailbox number.
3089 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3090 retrieve state for a particular bridge, where <name> is the conference name
3091 * app_directory now allows exiting at any time using the operator or pound key.
3092 * Voicemail now supports setting a locale per-mailbox.
3093 * Two new applications are provided for declining counting phrases in multiple
3094 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3096 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3098 * Voicemail now includes rdnis within msgXXXX.txt file.
3099 * ExternalIVR now supports IPv6 addresses.
3100 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3101 at https://wiki.asterisk.org/wiki/x/oQBB
3102 * ParkedCall and Park can now specify the parking lot to use.
3106 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3107 over SRV records associated with a specific service. From the CLI, type
3108 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3109 details on how these may be used.
3110 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3111 pitch of a channel's tx and rx audio streams.
3112 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3113 setting various connected line and redirecting party information.
3114 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3115 support ISDN subaddressing.
3116 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3117 * For DAHDI channels, the CHANNEL() dialplan function now allows
3118 the dialplan to request changes in the configuration of the active
3119 echo canceller on the channel (if any), for the current call only.
3122 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3124 The possible values are:
3126 on - normal mode (the echo canceller is actually reinitialized)
3128 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3130 voice - voice mode (returns from FAX mode, reverting the changes that
3131 were made when FAX mode was requested)
3132 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3133 and setting variables on the channel which created the current channel.
3134 Administrators should take care to avoid naming conflicts, when multiple
3135 channels are dialled at once, especially when used with the Local channel
3136 construct (which all could set variables on the master channel). Usage
3137 of the HASH() dialplan function, with the key set to the name of the slave
3138 channel, is one approach that will avoid conflicts.
3139 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3141 * func_odbc now allows multiple row results to be retrieved without using
3142 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3143 from the same query by using the name of the function which retrieved the
3144 first row as an argument to ODBC_FETCH().
3145 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3146 dialplan. This function returns the content of the received message.
3147 * Added REPLACE, which searches a given variable name for a set of characters,
3148 then either replaces them with a single character or deletes them.
3149 * Added PASSTHRU, which literally passes the same argument back as its return
3150 value. The intent is to be able to use a literal string argument to
3151 functions that currently require a variable name as an argument.
3152 * HASH-associated variables now can be inherited across channel creation, by
3153 prefixing the name of the hash at assignment with the appropriate number of
3154 underscores, just like variables.
3155 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3156 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3157 whether or not channels that are bridged to the current channel will be
3158 required to have secure signaling and/or media.
3159 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3160 the current channel has secure signaling and/or media.
3161 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3162 "no_media_path" option.
3163 Returns "0" if there is a B channel associated with the call.
3164 Returns "1" if no B channel is associated with the call. The call is either
3165 on hold or is a call waiting call.
3166 * Added option to dialplan function CDR(), the 'f' option
3167 allows for high resolution times for billsec and duration fields.
3168 * FILE() now supports line-mode and writing.
3169 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3170 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3174 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3175 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3176 and is set when a dynamic feature is triggered.
3177 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3178 to dynamically create a new parking lot matching the value this varible is
3180 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3181 features.conf that should be the base for dynamic parkinglots.
3182 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3183 parkinglot should have.
3184 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3185 parkinglot should have.
3186 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3191 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3192 timeout has expired.
3193 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3194 to the caller when an Agent's phone is ringing. This can be used to indicate
3195 to the caller that their call is about to be picked up, which is nice when
3196 one has been on hold for an extened period of time.
3197 * A new config option, penaltymemberslimit, has been added to queues.conf.
3198 When set this option will disregard penalty settings when a queue has too
3200 * A new option, 'I' has been added to both app_queue and app_dial.
3201 By setting this option, Asterisk will not update the caller with
3202 connected line changes or redirecting party changes when they occur.
3203 * A 'relative-periodic-announce' option has been added to queues.conf. When
3204 enabled, this option will cause periodic announce times to be calculated
3205 from the end of announcements rather than from the beginning.
3206 * The autopause option in queues.conf can be passed a new value, "all." The
3207 result is that if a member becomes auto-paused, he will be paused in all
3208 queues for which he is a member, not just the queue that failed to reach
3210 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3211 * The queue logger now allows events to optionally propagate to a file,
3212 even when realtime logging is turned on. Additionally, realtime logging
3213 supports sending the event arguments to 5 individual fields, although it
3214 will fallback to the previous data definition, if the new table layout is
3217 mISDN channel driver (chan_misdn) changes
3218 ----------------------------------------
3219 * Added display_connected parameter to misdn.conf to put a display string
3220 in the CONNECT message containing the connected name and/or number if
3221 the presentation setting permits it.
3222 * Added display_setup parameter to misdn.conf to put a display string
3223 in the SETUP message containing the caller name and/or number if the
3224 presentation setting permits it.
3225 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3226 indicate the dialplan settings are to be obtained from the asterisk
3228 * Made misdn.conf parameter callerid accept the "name" <number> format
3229 used by the rest of the system.
3230 * Made use the nationalprefix and internationalprefix misdn.conf
3231 parameters to prefix any received number from the ISDN link if that
3232 number has the corresponding Type-Of-Number. NOTE: This includes
3233 comparing the incoming call's dialed number against the MSN list.
3234 * Added the following new parameters: unknownprefix, netspecificprefix,
3235 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3236 received number from the ISDN link if that number has the corresponding
3238 * Added new dialplan application misdn_command which permits controlling
3239 the CCBS/CCNR functionality.
3240 * Added new dialplan function mISDN_CC which permits retrieval of various
3241 values from an active call completion record.
3242 * For PTP, you should manually send the COLR of the redirected-to party
3243 for an incomming redirected call if the incoming call could experience
3244 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3245 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3246 if the REDIRECTING(from-num) is not empty.
3247 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3248 option on all of the REDIRECTING statements before dialing the
3249 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3250 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3251 redirecting-to presentation (COLR) when it becomes available.
3252 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3255 thirdparty mISDN enhancements
3256 -----------------------------
3257 mISDN has been modified by Digium, Inc. to greatly expand facility message
3259 * Enhanced COLP support for call diversion and transfer.
3260 * CCBS/CCNR support.
3262 The latest modified mISDN v1.1.x based version is available at:
3263 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3264 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3266 Tagged versions of the modified mISDN code are available under:
3267 http://svn.digium.com/svn/thirdparty/mISDN/tags
3268 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3270 libpri channel driver (chan_dahdi) DAHDI changes
3271 -------------------------------------------
3272 * The channel variable PRIREDIRECTREASON is now just a status variable
3273 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3274 to read and alter the reason.
3275 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3276 redirected-to party for an incomming redirected call if the incoming call
3277 could experience further redirects. Just set the
3278 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3279 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3281 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3282 use the inhibit(i) option on all of the REDIRECTING statements before
3283 dialing the redirected-to party. You still have to set the
3284 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3285 will update the redirecting-to presentation (COLR) when it becomes available.
3286 * Added the ability to ignore calls that are not in a Multiple Subscriber
3287 Number (MSN) list for PTMP CPE interfaces.
3288 * Added dynamic range compression support for dahdi channels. It is
3289 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3290 * Added support for ISDN calling and called subaddress with partial support
3291 for connected line subaddress.
3292 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3293 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3294 to transfer a held call on disconnect similar to an analog phone.
3295 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3296 Will reroute/deflect an outgoing call when receive the message.
3297 Can use the DAHDISendCallreroutingFacility to send the message for the
3299 * Added standard location to add options to chan_dahdi dialing:
3300 Dial(DAHDI/g1[/extension[/options]])
3303 R Reverse charging indication
3304 * Added Reverse Charging Indication (Collect calls) send/receive option.
3305 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3306 Dial(DAHDI/g1/extension/R)
3307 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3308 (requires latest LibPRI)
3309 * Added ability to send/receive keypad digits in the SETUP message.
3310 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3311 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3312 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3313 (requires latest LibPRI)
3314 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3315 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3316 back into the same interface. Tromboned calls happen because of call routing,
3317 call deflection, call forwarding, and call transfer.
3318 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3319 * Added the ability to support call waiting calls. (The SETUP has no B channel
3321 * Added Malicious Call ID (MCID) event to the AMI call event class.
3322 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3324 Asterisk Manager Interface
3325 --------------------------
3326 * The Hangup action now accepts a Cause header which may be used to
3327 set the channel's hangup cause.
3328 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3329 to specify a separate .pem file to hold a private key. By default sslcert
3330 is used to hold both the public and private key.
3331 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3332 for options containing the 'tls' prefix. For example, 'sslenable' is now
3333 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3334 across all .conf files. All affected sample.conf files have been modified to
3335 reflect this change. Previous options such as 'sslenable' still work,
3336 but options with the 'tls' prefix are preferred.
3337 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3338 in a channel. (res_mutestream.so)
3339 * The configuration file manager.conf now supports a channelvars option, which
3340 specifies a list of channel variables to include in each channel-oriented
3342 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3343 and ExtraPriority to allow redirecting the second channel to a different
3344 location than the first.
3345 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3347 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3348 in a MixMonitor recording.
3349 * The 'iax2 show peers' output is now similar to the expected output of
3351 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3353 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3354 AOC-E messages on a channel.
3355 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3356 conform more closely to similar events.
3357 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3359 * Added optional parkinglot variable for park command.
3360 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3361 if CallerIDNum and CallerIDName headers are also present.
3363 Channel Event Logging
3364 ---------------------
3365 * A new interface, CEL, is introduced here. CEL logs single events, much like
3366 the AMI, but it differs from the AMI in that it logs to db backends much
3367 like CDR does; is based on the event subsystem introduced by Russell, and
3368 can share in all its benefits; allows multiple backends to operate like CDR;
3369 is specialized to event data that would be of concern to billing sytems,
3370 like CDR. Backends for logging and accounting calls have been produced,
3371 but a new CDR backend is still in development.
3375 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3376 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3377 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3378 * Multiple files and formats can now be specified in cdr_custom.conf.
3379 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3380 See configs/cdr_syslog.conf.sample for more information.
3381 * A 'sequence' field has been added to CDRs which can be combined with
3382 linkedid or uniqueid to uniquely identify a CDR.
3383 * Handling of billsec and duration field has changed. If your table definition
3384 specifies those fields as float,double or similar they will now be logged with
3385 microsecond accuracy instead of a whole integer.
3387 Calendaring for Asterisk
3388 ------------------------
3389 * A new set of modules were added supporing calendar integration with Asterisk.
3390 Dialplan functions for reading from and writing to calendars are included,
3391 as well as the ability to execute dialplan logic upon calendar event notifications.
3392 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3393 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3394 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
3395 2003 support does not support forms-based authentication).
3397 Call Completion Supplementary Services for Asterisk
3398 ---------------------------------------------------
3399 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
3400 DAHDI/ISDN supports call completion for the following switch types:
3401 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
3402 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
3404 Multicast RTP Support
3405 ---------------------
3406 * A new RTP engine and channel driver have been added which supports Multicast RTP.
3407 The channel driver can be used with the Page application to perform multicast RTP
3408 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
3409 Type can be either basic or linksys.
3410 Destination is the IP address and port for the RTP packets.
3411 Control address is specific to the linksys type and is used for sending the control
3412 packets unique to them.
3414 Security Events Framework
3415 -------------------------
3416 * Asterisk has a new C API for reporting security events. The module res_security_log
3417 sends these events to the "security" logger level. Currently, AMI is the only
3418 Asterisk component that reports security events. However, SIP support will be
3419 coming soon. For more information on the security events framework, see the
3420 "Asterisk Security Framework" section of the Asterisk wiki at
3421 https://wiki.asterisk.org/wiki/x/wgBQ
3422 * SIP support was added in Asterisk 10
3423 * This API now supports IPv6 addresses
3427 * A technology independent fax frontend (res_fax) has been added to Asterisk.
3428 * A spandsp based fax backend (res_fax_spandsp) has been added.
3429 * The app_fax module has been deprecated in favor of the res_fax module and
3430 the new res_fax_spandsp backend.
3431 * The SendFAX and ReceiveFAX applications now send their log messages to a
3432 'fax' logger level, instead of to the generic logger levels. To see these
3433 messages, the system's logger.conf file will need to direct the 'fax' logger
3434 level to one or more destinations; the logger.conf.sample file includes an
3435 example of how to do this. Note that if the 'fax' logger level is *not*
3436 directed to at least one destination, log messages generated by these
3437 applications will be lost, and that if the 'fax' logger level is directed to
3438 the console, the 'core set verbose' and 'core set debug' CLI commands will
3439 have no effect on whether the messages appear on the console or not.
3443 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
3444 Now, in order to enable transmitting silence during record the transmit_silence
3445 option should be used. transmit_silence_during_record remains a valid option, but
3446 defaults to the behavior of the transmit_silence option.
3447 * Addition of the Unit Test Framework API for managing registration and execution
3448 of unit tests with the purpose of verifying the operation of C functions.
3449 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
3450 XMPP text messages to the remote JID.
3451 * Modules.conf has a new option - "require" - that marks a module as critical for
3452 the execution of Asterisk.
3453 If one of the required modules fail to load, Asterisk will exit with a return
3455 * An 'X' option has been added to the asterisk application which enables #exec support.
3456 This allows #exec to be used in asterisk.conf.
3457 * jabber.conf supports a new option auth_policy that toggles auto user registration.
3458 * A new lockconfdir option has been added to asterisk.conf to protect the
3459 configuration directory (/etc/asterisk by default) during reloads.
3460 * The parkeddynamic option has been added to features.conf to enable the creation
3461 of dynamic parkinglots.
3462 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
3463 the reportalarms config option.
3464 * chan_dahdi supports dialing configuring and dialing by device file name.
3465 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3466 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3467 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3468 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3469 Handy for the above name-based syntax as it does not depend on
3470 initialization order.
3471 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3472 significant increase in performance (about 3X) for installations using this switchtype.
3473 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3474 AIS. For more information, please see the Distributed Device State section of the
3475 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3476 * The addition of G.719 pass-through support.
3477 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3478 during device configuration.
3479 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3480 have less than 3 lines on the LCD.
3481 * Realtime now supports database failover. See the sample extconfig.conf for details.
3482 * The addition of improved translation path building for wideband codecs. Sample
3483 rate changes during translation are now avoided unless absolutely necessary.
3484 * The addition of the res_stun_monitor module for monitoring and reacting to network
3485 changes while behind a NAT.
3486 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3487 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3488 These allow support for any Administration. Default is AT&T values.
3492 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3493 optionally accept a filename, to apply the setting only to the code generated from
3494 that source file when Asterisk was built. However, there are some modules in Asterisk
3495 that are composed of multiple source files, so this did not result in the behavior
3496 that users expected. In this version, 'core set debug' and 'core set verbose'
3497 can optionally accept *module* names instead (with or without the .so extension),
3498 which applies the setting to the entire module specified, regardless of which source
3499 files it was built from.
3500 * New 'manager show settings' command showing the current settings loaded from
3502 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3503 the channel hangup request to all channels.
3504 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3506 ------------------------------------------------------------------------------
3507 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3508 ------------------------------------------------------------------------------
3512 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3513 Snom phones use this for call pickup of extensions that the phone is
3515 * Added support for setting the domain in the URI for caller of an
3516 outbound call by using the SIPFROMDOMAIN channel variable.
3517 * Added a new configuration option "remotesecret" for authentication to
3518 remote services. For backwards compatibility, "secret" still has the
3519 same function as before, but now you can configure both a remote secret and a
3520 local secret for mutual authentication.
3521 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3522 the sound will be played to the target of an attended transfer
projects / asterisk/asterisk.git / blob