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 * Queue rules can now be stored in a database table, queue_rules. Unlike other
100 RealTime tables, the queue_rules table is only examined on module load or
101 module reload. A new general setting has been added to queuerules.conf,
102 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
103 RealTime for additional queue rules to parse. Note that both the file and
104 the database can be used as a provide of queue rules when 'realtime_rules'
107 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
108 There is no caching of RealTime queue rules.
112 * This module was deprecated and has been removed. Users of app_readfile
113 should use func_env's FILE function instead.
117 * The 'say' family of dialplan applications now support the Japanese
118 language. The 'language' parameter in say.conf now recognizes a setting of
119 'ja', which will enable Japanese language specific mechanisms for playing
120 back numbers, dates, and other items.
124 * This module was deprecated and has been removed. Users of app_saycountpl
125 should use the Say family of applications.
129 * The SetMusicOnHold dialplan application was deprecated and has been removed.
130 Users of the application should use the CHANNEL function's musicclass
135 * The WaitMusicOnHold dialplan application was deprecated and has been
136 removed. Users of the application should use MusicOnHold with a duration
141 * VoiceMail and VoiceMailMain now support the Japanese language. The
142 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
143 which will enable prompts to be played back using a Japanese grammatical
144 structure. Additional prompts are necessary for this functionality,
146 - jb-arimasu: there is
147 - jb-arimasen: there is not
148 - jb-oshitekudasai: please press
154 * Add the ability to specify multiple email addresses in configuration,
163 * This module was deprecated and has been removed. Users of cdr_sqlite
164 should use cdr_sqlite3_custom.
168 * Added the ability to support PostgreSQL application_name on connections.
169 This allows PostgreSQL to display the configured name in the
170 pg_stat_activity view and CSV log entries. This setting is configurable
171 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
179 * Added the ability to support PostgreSQL application_name on connections.
180 This allows PostgreSQL to display the configured name in the
181 pg_stat_activity view and CSV log entries. This setting is configurable
182 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
190 * SS7 support now requires libss7 v2.0 or later.
192 * Added SS7 support for connected line and redirecting.
194 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
197 * Added several SS7 config option parameters described in
198 chan_dahdi.conf.sample.
202 * This module was deprecated and has been removed. Users of chan_gtalk
203 should use chan_motif.
207 * This module was deprecated and has been removed. Users of chan_h323
208 should use chan_ooh323.
212 * This module was deprecated and has been removed. Users of chan_jingle
213 should use chan_motif.
217 * The SIPPEER dialplan function no longer supports using a colon as a
218 delimiter for parameters. The parameters for the function should be
219 delimited using a comma.
221 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
222 of the function should use the CHANNEL function instead.
230 * Added functional peeraccount support. Except for Queue, the
231 accountcode propagation is now consistently propagated to outgoing
232 channels before dialing. The channel accountcode can change from its
233 original non-empty value on channel creation for the following specific
234 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
235 originate method that can specify an accountcode value. Three, the
236 calling channel propagates its peeraccount or accountcode to the
237 outgoing channel's accountcode before dialing. The change has two
238 visible effects. One, local channels now cross accountcode and
239 peeraccount across the special bridge between the ;1 and ;2 channels
240 just like channels between normal bridges. Two, the
241 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
242 set the accountcode on the outgoing channel(s).
244 For Queue, an outgoing channel's non-empty accountcode will not change
245 unless explicitly set by CHANNEL(accountcode). The change has three
246 visible effects. One, local channels now cross accountcode and
247 peeraccount across the special bridge between the ;1 and ;2 channels
248 just like channels between normal bridges. Two, the queue member will
249 get an accountcode if it doesn't have one and one is available from the
250 calling channel's peeraccount. Three, accountcode propagation includes
251 local channel members where the accountcodes are propagated early
252 enough to be available on the ;2 channel.
256 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
257 These events are emitted whenever a device state or presence state change
258 occurs. The events are controlled by res_manager_device_state.so and
259 res_manager_presence_state.so. If the high frequency of these events is
260 problematic for you, do not load these modules.
262 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
263 work in basically the same way as the 'dialplan add extension' and
264 'dialplan remove extension' CLI commands respectively.
266 * New AMI action LoggerRotate reloads and rotates logger in the same manner
267 as CLI command 'logger rotate'
269 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
270 functionality of CLI commands 'fax show sessions', 'fax show session',
271 and fax show stats' respectively.
273 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
274 enable manager control over PRI debugging levels and file output.
276 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
277 endpoint as long as a default outbound endpoint is set. This also applies
278 to the equivalent CLI command (pjsip send notify)
280 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
281 that give information on Asterisk's attempts to qualify the endpoint.
285 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
286 and BRIDGE_EXIT events.
290 * Channel variables are now substituted in arguments passed to applications
291 run by using dynamic features.
295 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
296 Enabling PFS is attempted by default, and is dependent on the configuration
297 of the module using TLS.
298 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
299 specify a ECDHE cipher suite in sip.conf, for example:
300 tlscipher=AES128-SHA:DES-CBC3-SHA
301 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
302 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
303 default dh2048.pem - see
304 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
305 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
306 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
307 Consider re-ordering your cipher suites in the respective configuration
309 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
310 will use PFS when offered by the client. Clients which do not offer PFS
311 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
319 * The JACK_HOOK function now supports audio with a sample rate higher than
328 * Added the ability to support PostgreSQL application_name on connections.
329 This allows PostgreSQL to display the configured name in the
330 pg_stat_activity view and CSV log entries. This setting is configurable
331 for res_config_pgsql via the dbappname configuration setting in
334 res_pjsip_outbound_publish
336 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
337 PUBLISH requests for specific event packages to another SIP User Agent.
341 * The publish/subscribe core module has been updated to support RFC 4662
342 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
343 Resource lists are configured in pjsip.conf under a new object type,
344 resource_list. Resource lists can contain either message-summary or presence
345 events, and can be composed of specific resources that provide the event or
346 other resource lists.
348 * Inbound publication support is provided by a new object, inbound-publication.
349 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
350 resource. Which events are accepted is constructed dynamically; see
351 res_pjsip_publish_asterisk for more information.
353 res_pjsip_publish_asterisk
355 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
356 Asterisk information to other Asterisk servers. This module is intended only
357 for Asterisk to Asterisk exchanges of information. Currently, this includes
358 both mailbox state and device state information.
361 ------------------------------------------------------------------------------
362 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
363 ------------------------------------------------------------------------------
367 * Stored recordings now support a new operation, copy. This will take an
368 existing stored recording and copy it to a new location in the recordings
371 * LiveRecording objects now have three additional fields that can be reported
372 in a RecordingFinished ARI event:
373 - total_duration: the duration of the recording
374 - talking_duration: optional. The duration of talking detected in the
375 recording. This is only available if max_silence_seconds was specified
376 when the recording was started.
377 - silence_duration: optional. The duration of silence detected in the
378 recording. This is only available if max_silence_seconds was specified
379 when the recording was started.
380 Note that all duration values are reported in seconds.
382 * Users of ARI can now send and receive out of call text messages. Messages
383 can be sent directly to a particular endpoint, or can be sent to the
384 endpoints resource directly and inferred from the URI scheme. Text
385 messages are passed to ARI clients as TextMessageReceived events. ARI
386 clients can choose to receive text messages by subscribing to the particular
387 endpoint technology or endpoints that they are interested in.
389 * The applications resource now supports subscriptions to all endpoints of
390 a particular channel technology. For example, subscribing to an eventSource
391 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
395 * The endpoint configuration object now supports 'accountcode'. Any channel
396 created for an endpoint with this setting will have its accountcode set
397 to the specified value.
401 * A new module, res_hep_rtcp, has been added that will forward RTCP call
402 statistics to a HEP capture server. See res_hep for more information.
406 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
407 unconditionally inhereted through masquerades. As a side benefit, more
408 than one audiohook of a given type may persist through a masquerade now.
410 ------------------------------------------------------------------------------
411 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
412 ------------------------------------------------------------------------------
416 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
417 connect with an incoming caller after being alerted to the presence
418 of the incoming caller. The most likely reason this would happen is
419 the agent did not acknowledge the call in time.
423 * New events have been added for the TALK_DETECT function. When the function
424 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
425 emitted to connected AMI clients indicating the start/stop of talking on
430 * New event models have been aded for the TALK_DETECT function. When the
431 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
432 events will be emitted to connected WebSockets subscribed to the channel,
433 indicating the start/stop of talking on the channel.
437 * A new function, TALK_DETECT, has been added. When set on a channel, this
438 fucntion causes events indicating the starting/stoping of talking on said
439 channel to be emitted to both AMI and ARI clients.
441 ------------------------------------------------------------------------------
442 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
443 ------------------------------------------------------------------------------
447 * A new Playback URI 'tone' has been added. Tones are specified either as
448 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
449 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
450 URIs in that they must be stopped manually and will continue to occupy
451 a channel's ARI control queue until they are stopped. They also can not
452 be rewound or fastforwarded.
454 * User events can now be generated from ARI. Events can be signalled with
455 arbitrary json variables, and include one or more of channel, bridge, or
456 endpoint snapshots. An application must be specified which will receive
457 the event message (other applications can subscribe to it). The message
458 will also be delivered via AMI provided a channel is attached. Dialplan
459 generated user event messages are still transmitted via the channel, and
460 will only be received by a stasis application they are attached to or if
461 the channel is subscribed to.
465 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
466 fields for prohibited callingpres information. Values are legacy, no, and
467 yes. By default, legacy is used.
468 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
469 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
470 headers are appended to outbound SIP messages just as they are with
471 allowed callingpres values, but data about the remote party's identity is
473 When sendrpid=rpid, only the remote party's domain is anonymized.
474 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
475 headers are not sent.
476 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
477 party information in tact even for prohibited callingpres information.
478 In the case of PAI, a Privacy: id header will be appended for prohibited
479 calling information to communicate that the private information should
480 not be relayed to untrusted parties.
484 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
485 which can be used to announce the parked call's location to an arbitrary
486 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
487 parties in a one to one bridge, 'TimeoutChannel' is treated as having
488 parked 'Channel' like with the Park Call DTMF feature and will receive
489 announcements prior to being hung up.
491 ------------------------------------------------------------------------------
492 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
493 ------------------------------------------------------------------------------
497 * Record application now has an option 'o' which allows 0 to act as an exit
498 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
501 --------------------------
502 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
503 as the chanprefix parameter if the 'u' option is specified.
506 --------------------------
507 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
508 conference user menus.
510 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
511 menus, bridge settings, and user settings that have been applied by the
512 CONFBRIDGE dialplan function.
514 * The ConfBridge dialplan application now sets a channel variable,
515 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
516 how a channel exited the conference.
518 * Added conference user option 'announce_join_leave_review'. This option
519 implies 'announce_join_leave' with the added effect that the user will
520 be asked if they want to confirm or re-record the recording of their
521 name when entering the conference
524 --------------------------
525 * At exit, the Directory application now sets a channel variable
526 DIRECTORY_RESULT to one of the following based on the reason for exiting:
527 OPERATOR user requested operator by pressing '0' for operator
528 ASSISTANT user requested assistant by pressing '*' for assistant
529 TIMEOUT user pressed nothing and Directory stopped waiting
530 HANGUP user's channel hung up
531 SELECTED user selected a user from the directory and is routed
532 USEREXIT user pressed '#' from the selection prompt to exit
533 FAILED directory failed in a way that wasn't accounted for. Dang.
537 * Monitor() - A new option, B(), has been added that will turn on a periodic
538 beep while the call is being recorded.
541 --------------------------
542 * MusicOnHold streams (all modes other than "files") now support wide band
546 --------------------------
547 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
548 and for the channel executing Page respectively.
551 --------------------------
552 * PickupChan now accepts channel uniqueids of channels to pickup.
555 --------------------------
556 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
557 to 'true' (case insensitive), then any Say application (SayNumber,
558 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
559 anticipate DTMF. If DTMF is received, these applications will behave like
560 the background application and jump to the received extension once a match
561 is established or after a short period of inactivity.
564 -------------------------
565 * A new function, MIXMONITOR, has been added to allow access to individual
566 instances of MixMonitor on a channel.
568 * A new option, B(), has been added that will turn on a periodic beep while the
569 call is being recorded.
573 -------------------------
576 -------------------------
577 * TEL URI support for inbound INVITE requests has been added. chan_sip will
578 now handle TEL schemes in the Request and From URIs. The phone-context in
579 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
584 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
585 the new AST_SORCERY diaplan function.
587 * Core Show Locks output now includes Thread/LWP ID if the platform
588 supports this feature.
590 * New "logger add channel" and "logger remove channel" CLI commands have
591 been added to allow creation and deletion of dynamic logger channels
592 without configuration changes. These dynamic logger channels will only
593 exist until the next restart of asterisk.
597 * The live recording object on recording events now contains a target_uri
598 field which contains the URI of what is being recorded.
600 * The bridge type used when creating a bridge is now a comma separated list of
601 bridge properties. Valid options are: mixing, holding, dtmf_events, and
604 * A channelId can now be provided when creating a channel, either in the
605 uri (POST channels/my-channel-id) or as query parameter. A local channel
606 will suffix the second channel id with ';2' unless provided as query
607 parameter otherChannelId.
609 * A bridgeId can now be provided when creating a bridge, either in the uri
610 (POST bridges/my-bridge-id) or as a query parameter.
612 * A playbackId can be provided when starting a playback, either in the uri
613 (POST channels/my-channel-id/play/my-playback-id /
614 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
616 * A snoop channel can be started with a snoopId, in the uri or query.
620 * Originate now takes optional parameters ChannelId and OtherChannelId,
621 used to set the UniqueId on creation. The other id is assigned to the
622 second channel when dialing LOCAL, or defaults to appending ;2 if only
623 the single Id is given.
625 * The Mixmonitor action now has a "Command" header that can be used to
626 indicate a post-process command to run once recording finishes.
630 * A new set of Alembic scripts has been added for CDR tables. This will create
631 a 'cdr' table with the default schema that Asterisk expects.
636 * A new function was added: PERIODIC_HOOK. This allows running a periodic
637 dialplan hook on a channel. Any audio generated by this hook will be
638 injected into the call.
646 * A new module, res_hep, has been added, that acts as a generic packet
647 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
648 It can be configured via hep.conf. Other modules can use res_hep to send
649 message traffic to a HEP capture server.
653 * A new module, res_hep_pjsip, has been added that will forward PJSIP
654 message traffic to a HEP capture server. See res_hep for more
659 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
660 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
662 * Added the following new CLI commands:
663 - "pjsip show contacts" - list all current PJSIP contacts.
664 - "pjsip show contact" - show specific information about a current PJSIP
666 - "pjsip show channel" - show detailed information about a PJSIP channel.
670 * A new module, res_pjsip_multihomed handles situations where the system
671 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
672 determines which interface should be used during message sending.
674 res_pjsip_pidf_digium_body_supplement
676 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
677 request body formatting for presence support in Digium phones.
679 res_pjsip_send_to_voicemail
681 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
682 particular headers to transfer a PJSIP channel directly to a particular
683 extension that has VoiceMail. This is intended to be used with Digium
684 phones that support this feature.
686 res_pjsip_outbound_registration
688 * A new CLI command has been added: "pjsip show registrations", which lists
689 all configured PJSIP registrations
692 ------------------------------------------------------------------------------
693 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
694 ------------------------------------------------------------------------------
698 * Added a new module that provides AMI control over MWI within Asterisk,
699 res_mwi_external_ami. Note that this module depends on res_mwi_external;
700 for more information on enabling this module, see res_mwi_external.
701 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
702 the MWIGet/MWIGetComplete events.
704 * The DialStatus field in the DialEnd event can now contain additional
705 statuses that convey how the dial operation terminated. This includes
706 ABORT, CONTINUE, and GOTO.
708 * AMI will now emit security events. A new class authorization has been
709 added in manager.conf for the security events, 'security'. The new events
711 - FailedACL - raised when a request violates an ACL check
712 - InvalidAccountID - raised when a request fails an authentication
713 check due to an invalid account ID
714 - SessionLimit - raised when a request fails due to exceeding the
715 number of allowed concurrent sessions for a service
716 - MemoryLimit - raised when a request fails due to an internal memory
718 - LoadAverageLimit - raised when a request fails because a configured
719 load average limit has been reached
720 - RequestNotAllowed - raised when a request is not allowed by
722 - AuthMethodNotAllowed - raised when a request used an authentication
723 method not allowed by the service
724 - RequestBadFormat - raised when a request is received with bad formatting
725 - SuccessfulAuth - raised when a request successfully authenticates
726 - UnexpectedAddress - raised when a request has a different source address
727 then what is expected for a session already in progress with a service
728 - ChallengeResponseFailed - raised when a request's attempt to authenticate
729 has been challenged, and the request failed the authentication challenge
730 - InvalidPassword - raised when a request provides an invalid password
731 during an authentication attempt
732 - ChallengeSent - raised when an Asterisk service send an authentication
733 challenge to a request
734 - InvalidTransport - raised when a request attempts to use a transport not
735 allowed by the Asterisk service
737 * Bridge related events now have two additional fields: BridgeName and
738 BridgeCreator. BridgeName is a descriptive name for the bridge;
739 BridgeCreator is the name of the entity that created the bridge. This
740 affects the following events: ConfbridgeStart, ConfbridgeEnd,
741 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
742 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
743 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
747 * The Bridge data model now contains the additional fields 'name' and
748 'creator'. The 'name' field conveys a descriptive name for the bridge;
749 the 'creator' field conveys the name of the entity that created the bridge.
750 This affects all responses to HTTP requests that return a Bridge data model
751 as well as all event derived data models that contain a Bridge data model.
752 The POST /bridges operation may now optionally specify a name to give to
753 the bridge being created.
755 * Added a new ARI resource 'mailboxes' which allows the creation and
756 modification of mailboxes managed by external MWI. Modules res_mwi_external
757 and res_stasis_mailbox must be enabled to use this resource. For more
758 information on external MWI control, see res_mwi_external.
760 * Added new events for externally initiated transfers. The event
761 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
762 of a bridge in the ARI controlled application to the dialplan; the
763 BridgeAttendedTransfer event is raised when a channel initiates an
764 attended transfer of a bridge in the ARI controlled application to the
767 * Channel variables may now be specified as a body parameter to the
768 POST /channels operation. The 'variables' key in the JSON is interpreted
769 as a sequence of key/value pairs that will be added to the created channel
770 as channel variables. Other parameters in the JSON body are treated as
771 query parameters of the same name.
775 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
776 automatically handled by the HTTP server if a request is received with a
777 Transfer-Encoding type of "chunked".
781 * Path support has been added with the 'support_path' option in registration
784 * A 'debug' option has been added to the globals section that will allow
785 sip messages to be logged.
787 * A 'set_var' option has been added to endpoints that will automatically
788 set the desired variable(s) on a channel created for that endpoint.
790 * Several new tables and columns have been added to the realtime schema for
791 the res_pjsip related modules. See the UPGRADE.txt notes for updating
796 * A new module, res_mwi_external, has been added to Asterisk. This module
797 acts as a base framework that other modules can build on top of to allow
798 an external system to control MWI within Asterisk. For implementations
799 that make use of res_mwi_external, see res_mwi_external_ami and
800 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
801 that may produce MWI themselves, such as app_voicemail. res_mwi_external
802 and other modules that depend on it cannot be built or loaded with
803 app_voicemail present.
807 * DNS functionality will now automatically be enabled if the system configured
808 nameservers can be retrieved. If the system configured nameservers can not be
809 retrieved the functionality will resort to using system resolution. Functionalty
810 such as SRV records and failover will not be available if system resolution
813 ------------------------------------------------------------------------------
814 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
815 ------------------------------------------------------------------------------
820 Asterisk 12 is a standard release of the Asterisk project. As such, the
821 focus of development for this release was on core architectural changes and
822 major new features. This includes:
823 * A more flexible bridging core based on the Bridging API
824 * A new internal message bus, Stasis
825 * Major standardization and consistency improvements to AMI
826 * Addition of the Asterisk RESTful Interface (ARI)
827 * A new SIP channel driver, chan_pjsip
828 In addition, as the vast majority of bridging in Asterisk was migrated to the
829 Bridging API used by ConfBridge, major changes were made to most of the
830 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
832 Specifications have been written for the affected interfaces. These
833 specifications are available on the Asterisk wiki:
834 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
835 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
836 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
838 It is *highly* recommended that anyone migrating to Asterisk 12 read the
839 information regarding its release both in this file and in the accompanying
840 UPGRADE.txt file. More detailed information on the major changes can be found
841 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
846 * Added build option DISABLE_INLINE. This option can be used to work around a
847 bug in gcc. For more information, see
848 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
850 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
851 the CHANNEL_TRACE build option were incompatible with the new bridging
854 * Asterisk now optionally uses libxslt to improve XML documentation generation
855 and maintainability. If libxslt is not available on the system, some XML
856 documentation will be incomplete.
858 * Asterisk now depends on libjansson. If a package of libjansson is not
859 available on your distro, please see http://www.digip.org/jansson/.
861 * Asterisk now depends on libuuid and, optionally, uriparser. It is
862 recommended that you install uriparser, even if it is optional.
864 * The new SIP stack and channel driver uses a particular version of PJSIP.
865 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
866 configuring and installing PJSIP for usage with Asterisk.
868 * Optional API was re-implemented to be more portable, and no longer requires
869 weak reference support from the compiler. The build option OPTIONAL_API may
870 be disabled to disable Optional API support.
877 * Along with AgentRequest, this application has been modified to be a
878 replacement for chan_agent. The act of a channel calling the AgentLogin
879 application places the channel into a pool of agents that can be
880 requested by the AgentRequest application. Note that this application, as
881 well as all other agent related functionality, is now provided by the
882 app_agent_pool module. See chan_agent and AgentRequest for more information.
884 * This application no longer performs agent authentication. If authentication
885 is desired, the dialplan needs to perform this function using the
886 Authenticate or VMAuthenticate application or through an AGI script before
889 * If this application is called and the agent is already logged in, the
890 dialplan will continue exection with the AGENT_STATUS channel variable set
891 to ALREADY_LOGGED_IN.
893 * The agents.conf schema has changed. Rather than specifying agents on a
894 single line in comma delineated fashion, each agent is defined in a separate
895 context. This allows agents to use the power of context templates in their
898 * A number of parameters from agents.conf have been removed. This includes
899 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
900 urlprefix, and savecallsin. These options were obsoleted by the move from
901 a channel driver model to the bridging/application model provided by
906 * A new application, this will request a logged in agent from the pool and
907 bridge the requested channel with the channel calling this application.
908 Logged in agents are those channels that called the AgentLogin application.
909 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
910 application will be set with an appropriate error value.
914 * This application has been removed. It was a holdover from when
915 AgentCallbackLogin was removed.
919 * Added support for additional Ademco DTMF signalling formats, including
920 Express 4+1, Express 4+2, High Speed and Super Fast.
922 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
923 call time, in milliseconds, to run the application.
925 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
926 maximum number of times to retry the call.
928 * Added a new configuration option answait. If set, the AlarmReceiver
929 application will wait the number of milliseconds specified by answait
930 after the channel has answered. Valid values range between 500
931 milliseconds and 10000 milliseconds.
933 * Added configuration option no_group_meta. If enabled, grouping of metadata
934 information in the AlarmReceiver log file will be skipped.
938 * It is now no longer possible to bypass updating the CDR on the channel
939 when answering. CDRs reflect the state of the channel and will always
940 reflect the time they were Answered.
944 * A new application in Asterisk, this will place the calling channel
945 into a holding bridge, optionally entertaining them with some form of
946 media. Channels participating in a holding bridge do not interact with
947 other channels in the same holding bridge. Optionally, however, a channel
948 may join as an announcer. Any media passed from an announcer channel is
949 played to all channels in the holding bridge. Channels leave a holding
950 bridge either when an optional timer expires, or via the ChannelRedirect
951 application or AMI Redirect action.
955 * All participants in a bridge can now be kicked out of a conference room
956 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
957 command, i.e., 'confbridge kick <conference> all'
959 * CLI output for the 'confbridge list' command has been improved. When
960 displaying information about a particular bridge, flags will now be shown
961 for the participating users indicating properties of that user.
963 * The ConfbridgeList event now contains the following fields: WaitMarked,
964 EndMarked, and Waiting. This displays additional properties about the
965 user's profile, as well as whether or not the user is waiting for a
966 Marked user to enter the conference.
968 * Added a new option for conference recording, record_file_append. If enabled,
969 when the recording is stopped and then re-started, the existing recording
970 will be used and appended to.
972 * ConfBridge now has the ability to set the language of announcements to the
973 conference. The language can be set on a bridge profile in confbridge.conf
974 or by the dialplan function CONFBRIDGE(bridge,language)=en.
978 * The channel variable CPLAYBACKSTATUS may now return the value
979 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
980 such as AMI. See the AMI action ControlPlayback for more information.
984 * Added the 'a' option, which allows the caller to enter in an additional
985 alias for the user in the directory. This option must be used in conjunction
986 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
987 specified in voicemail.conf.
991 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
992 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
993 containing the unique ID of the bridge that the channel happens to be in.
997 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
998 for more information.
1000 * Variables are no longer purged from the original CDR. See the 'v' option for
1003 * The 'A' option has been removed. The Answer time on a CDR is never updated
1006 * The 'd' option has been removed. The disposition on a CDR is a function of
1007 the state of the channel and cannot be altered.
1009 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1010 of the state of the respective channels involved in the CDR and cannot be
1013 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1014 such that the start time and, if applicable, the answer time was updated.
1015 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1016 'r' option now triggers the Reset, setting the start time (and answer time
1017 if applicable) to the current time. Note that the 'a' option still sets
1018 the answer time to the current time if the channel was already answered.
1020 * The 's' option has been removed. A variable can be set on the original CDR
1021 if desired using the CDR function, and removed from a forked CDR using the
1024 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1025 longer applies in the CDR engine.
1027 * The 'v' option now prevents the copy of the variables from the original CDR
1028 to the forked CDR. Previously the variables were always copied but were
1029 removed from the original. This was changed as removing variables from a CDR
1030 can have unintended side effects - this option allows the user to prevent
1031 propagation of variables from the original to the forked without modifying
1036 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1037 function to a channel joining a conference. Some channel drivers that vary
1038 the number of audio samples in a voice frame will experience significant
1039 quality problems if a denoiser is attached to the channel; this option gives
1040 them the ability to remove the denoiser without having to unload func_speex.
1044 * The 'b' option now includes conferences as well as sounds played to the
1047 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1048 running during a transfer. If a MixMonitor is started on a channel,
1049 the MixMonitor will continue to record the audio passing through the
1050 channel even in the presence of transfers.
1054 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1057 * While the NoCDR application will prevent CDRs for a channel from being
1058 propagated to registered CDR backends, it will not prevent that data from
1059 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1060 function that enables CDRs on a channel will restore those records that have
1061 not yet been finalized.
1065 * The app_parkandannounce module has been removed. The application
1066 ParkAndAnnounce is now provided by the res_parking module. See the
1067 res_parking changes for more information.
1071 * Added queue available hint. The hint can be added to the dialplan using the
1072 following syntax: exten,hint,Queue:{queue_name}_avail
1073 For example, if the name of the queue is 'markq':
1074 exten => 8501,hint,Queue:markq_avail
1075 This will report 'InUse' if there are no logged in agents or no free agents.
1076 It will report 'Idle' when an agent is free.
1078 * Queues now support a hint for member paused state. The hint uses the form
1079 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1080 are the name of the queue and the name of the member to subscribe to,
1081 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1082 Members will show as In Use when paused.
1084 * The configuration options eventwhencalled and eventmemberstatus have been
1085 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1086 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1087 sent. The "Variable" fields will also no longer exist on the Agent* events.
1088 These events can be filtered out from a connected AMI client using the
1089 eventfilter setting in manager.conf.
1091 * The queue log now differentiates between blind and attended transfers. A
1092 blind transfer will result in a BLINDTRANSFER message with the destination
1093 context and extension. An attended transfer will result in an
1094 ATTENDEDTRANSFER message. This message will indicate the method by which
1095 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1096 for running an application on a bridge or channel, or "LINK" for linking
1097 two bridges together with local channels. The queue log will also now detect
1098 externally initiated blind and attended transfers and record the transfer
1101 * When performing queue pause/unpause on an interface without specifying an
1102 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1103 least one member of any queue exists for that interface.
1105 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1106 for realtime queue log entries.
1110 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1111 CDRs when they were previously disabled on a channel.
1113 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1114 backends occurs on an as-needed basis in order to preserve linkedid
1115 propagation and other needed behavior.
1119 * A new application, this is similar to SayAlpha except that it supports
1120 case sensitive playback of the specified characters. For example,
1121 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1125 * This application is deprecated in favor of CHANNEL(amaflags).
1129 * The SendDTMF application will now accept 'W' as valid input. This will cause
1130 the application to delay one second while streaming DTMF.
1134 * A new application in Asterisk 12, this hands control of the channel calling
1135 the application over to an external system. Currently, external systems
1136 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1140 * UserEvent will now handle duplicate keys by overwriting the previous value
1141 assigned to the key.
1143 * In addition to AMI, UserEvent invocations will now be distributed to any
1144 interested Stasis applications.
1148 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1149 system as mailbox@context. The rest of the system cannot add @default
1150 to mailbox identifiers for app_voicemail that do not specify a context
1151 any longer. It is a mailbox identifier format that should only be
1152 interpreted by app_voicemail.
1154 * The voicemail.conf configuration file now has an 'alias' configuration
1155 parameter for use with the Directory application. The voicemail realtime
1156 database table schema has also been updated with an 'alias' column.
1161 * Pass through support has been added for both VP8 and Opus.
1163 * Added format attribute negotiation for the Opus codec. Format attribute
1164 negotiation is provided by the res_format_attr_opus module.
1169 * Masquerades as an operation inside Asterisk have been effectively hidden
1170 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1171 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1172 dropping of frame/audio hooks, and other internal implementation details
1173 that users had to deal with. This fundamental change has large implications
1174 throughout the changes documented for this version. For more information
1175 about the new core architecture of Asterisk, please see the Asterisk wiki.
1177 * Multiple parties in a bridge may now be transferred. If a participant in a
1178 multi-party bridge initiates a blind transfer, a Local channel will be used
1179 to execute the dialplan location that the transferer sent the parties to. If
1180 a participant in a multi-party bridge initiates an attended transfer,
1181 several options are possible. If the attended transfer results in a transfer
1182 to an application, a Local channel is used. If the attended transfer results
1183 in a transfer to another channel, the resulting channels will be merged into
1186 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1187 driver specific. If the channel variable is set on the transferrer channel,
1188 the sound will be played to the target of an attended transfer.
1190 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1191 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1192 listed. Any more peers in the bridge will not be included in the list.
1193 BRIDGEPEER is not valid in holding bridges like parking since those channels
1194 do not talk to each other even though they are in a bridge.
1196 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1197 and will contain a value if the BRIDGEPEER's channel driver supports it.
1199 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1200 was responsible for an attended transfer in a similar fashion to
1203 * Modules using the Configuration Framework or Sorcery must have XML
1204 configuration documentation. This configuration documentation is included
1205 with the rest of Asterisk's XML documentation, and is accessible via CLI
1206 commands. See the CLI changes for more information.
1208 AMI (Asterisk Manager Interface)
1210 * Major changes were made to both the syntax as well as the semantics of the
1211 AMI protocol. In particular, AMI events have been substantially improved
1212 in this version of Asterisk. For more information, please see the AMI
1213 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1215 * AMI events that reference a particular channel or bridge will now always
1216 contain a standard set of fields. When multiple channels or bridges are
1217 referenced in an event, fields for at least some subset of the channels
1218 and bridges in the event will be prefixed with a descriptive name to avoid
1219 name collisions. See the AMI event documentation on the Asterisk wiki for
1222 * The CLI command 'manager show commands' no longer truncates command names
1223 longer than 15 characters and no longer shows authorization requirement
1224 for commands. 'manager show command' now displays the privileges needed
1225 for using a given manager command instead.
1227 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1228 peer in its response if the peer has a subscribe context set.
1230 * The SIPqualifypeer action now acknowledges the request once it has
1231 established that the request is against a known peer. It also issues a new
1232 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1234 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1235 specifies the duration of the digit to be played, in milliseconds.
1237 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1238 updates when changes occur instead of requiring the use of pollmailboxes.
1240 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1241 AMI client to manipulate audio currently being played back on a channel. The
1242 supported operations depend on the application being used to send audio to
1243 the channel. When the audio playback was initiated using the ControlPlayback
1244 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1245 stopped, restarted, reversed, or skipped forward. When initiated by other
1246 mechanisms (such as the Playback application), the audio can be stopped,
1247 reversed, or skipped forward.
1249 * Channel related events now contain a snapshot of channel state, adding new
1250 fields to many of these events.
1252 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1253 in a future release. Please use the common 'Exten' field instead.
1255 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1256 fields. The channel state fields will come before the body fields.
1258 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1259 'UnParkedCall' have changed significantly in the new res_parking module.
1261 The 'Channel' and 'From' headers are gone. For the channel that was parked
1262 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1263 has a number of fields associated with it. The old 'Channel' header relayed
1264 the same data as the new 'ParkeeChannel' header.
1266 The 'From' field was ambiguous and changed meaning depending on the event.
1267 for most of these, it was the name of the channel that parked the call
1268 (the 'Parker'). There is no longer a header that provides this channel name,
1269 however the 'ParkerDialString' will contain a dialstring to redial the
1270 device that parked the call.
1272 On UnParkedCall events, the 'From' header would instead represent the
1273 channel responsible for retrieving the parkee. It receives a channel
1274 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1277 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1279 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1280 fashion has changed the field names 'StartExten' and 'StopExten' to
1281 'StartSpace' and 'StopSpace' respectively.
1283 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1284 manager.conf has been removed.
1286 * Channel Variables conveyed with a channel no longer contain the name of the
1287 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1288 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1289 event, the various ChanVariable fields will contain a suffix that specifies
1290 which channel they correspond to.
1292 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1293 event always conveys the AMI event for a particular channel.
1295 * All 'Reload' events have been consolidated into a single event type. This
1296 event will always contain a Module field specifying the name of the module
1297 and a Status field denoting the result of the reload. All modules now issue
1298 this event when being reloaded.
1300 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1301 fail to receive this event due to being connected after modules have loaded.
1302 AMI connections that want to know when Asterisk is ready should listen for
1303 the 'FullyBooted' event.
1305 * app_fax now sends the same send fax/receive fax events as res_fax. The
1306 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1307 now the 'ReceiveFAX' event.
1309 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1310 'MusicOnHoldStop'. The sub type field has been removed.
1312 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1313 carrier for another protocol.
1315 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1316 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1317 to the specific channel. 'Both' may be specified to play a tone to both
1318 channels. The old 'yes' option is still accepted as a way of playing the
1319 tone to Channel2 only.
1321 * The AMI 'Status' response event to the AMI Status action replaces the
1322 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1323 indicate what bridge the channel is currently in.
1325 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1326 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1329 * The AMI events in app_queue have been made more consistent with each other.
1330 Events that reference channels (QueueCaller* and Agent*) will show
1331 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1332 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1334 * The 'MCID' AMI event now publishes a channel snapshot when available and
1335 its non-channel-snapshot parameters now use either the "MCallerID" or
1336 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1337 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1338 parameters in the channel snapshot.
1340 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1341 'AgentLogin' and 'AgentLogoff' respectively.
1343 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1344 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1346 * 'ChannelUpdate' events have been removed.
1348 * All AMI events now contain a 'SystemName' field, if available.
1350 * Local channel optimization is now conveyed in two events:
1351 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1352 when the Local channel driver begins attempting to optimize itself out of
1353 the media path; the End event is sent after the channel halves have
1354 successfully optimized themselves out of the media path.
1356 * Local channel information in events is now prefixed with 'LocalOne' and
1357 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1358 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1359 and 'LocalOptimizationEnd' events.
1361 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1362 account. When set in the general context, it will act as the default
1363 setting for defined accounts.
1365 * The 'BridgeAction' event was removed. It technically added no value, as the
1366 Bridge Action already receives confirmation of the bridge through a
1367 successful completion Event.
1369 * The 'BridgeExec' events were removed. These events duplicated the events that
1370 occur in the Briding API, and are conveyed now through BridgeCreate,
1371 BridgeEnter, and BridgeLeave events.
1373 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1374 previous versions. They now report all SR/RR packets sent/received, and
1375 have been restructured to better reflect the data sent in a SR/RR. In
1376 particular, the event structure now supports multiple report blocks.
1378 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1379 raised when a blind transfer/attended transfer completes successfully.
1380 They contain information about the transfer that just completed, including
1381 the location of the transfered channel.
1383 * Added a 'security' class to AMI which outputs the required fields for
1384 security messages similar to the log messages from res_security_log
1386 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1387 that describes the status value in a human readable string.
1389 CDR (Call Detail Records)
1391 * Significant changes have been made to the behavior of CDRs. The CDR engine
1392 was effectively rewritten and built on the Stasis message bus. For a full
1393 definition of CDR behavior in Asterisk 12, please read the specification
1394 on the Asterisk wiki (wiki.asterisk.org).
1396 * CDRs will now be created between all participants in a bridge. For each
1397 pair of channels in a bridge, a CDR is created to represent the path of
1398 communication between those two endpoints. This lets an end user choose who
1399 to bill for what during bridge operations with multiple parties.
1401 * The duration, billsec, start, answer, and end times now reflect the times
1402 associated with the current CDR for the channel, as opposed to a cumulative
1403 measurement of all CDRs for that channel.
1405 * When a CDR is dispatched, user defined CDR variables from both parties are
1406 included in the resulting CDR. If both parties have the same variable, only
1407 the Party A value is provided.
1409 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1410 information regarding the CDR engine is logged as verbose messages. This
1411 option should only be used if the behavior of the CDR engine needs to be
1414 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1415 normally configured in cdr.conf.
1417 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1418 specified, this command provides a summary of the channels with CDR
1419 information and their statistics. When {channel} is specified, it shows
1420 detailed information about all records associated with {channel}.
1422 CEL (Channel Event Logging)
1424 * CEL has undergone significant rework in Asterisk 12, and is now built on the
1425 Stasis message bus. Please see the specification for CEL on the Asterisk
1426 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
1429 * The 'extra' field of all CEL events that use it now consists of a JSON blob
1430 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
1432 * BLINDTRANSFER events now report the transferee bridge unique
1433 identifier, extension, and context in a JSON blob as the extra string
1434 instead of the transferee channel name as the peer.
1436 * ATTENDEDTRANSFER events now report the peer as NULL and additional
1437 information in the 'extra' string as a JSON blob. For transfers that occur
1438 between two bridged channels, the 'extra' JSON blob contains the primary
1439 bridge unique identifier, the secondary channel name, and the secondary
1440 bridge unique identifier. For transfers that occur between a bridged channel
1441 and a channel running an app, the 'extra' JSON blob contains the primary
1442 bridge unique identifier, the secondary channel name, and the app name.
1444 * LOCAL_OPTIMIZE events have been added to convey local channel
1445 optimizations with the record occurring for the semi-one channel and
1446 the semi-two channel name in the peer field.
1448 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
1449 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
1450 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
1451 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
1452 regardless of whether or not that bridge happens to contain multiple
1457 * When compiled with '--enable-dev-mode', the astobj2 library will now add
1458 several CLI commands that allow for inspection of ao2 containers that
1459 register themselves with astobj2. The CLI commands are 'astobj2 container
1460 dump', 'astobj2 container stats', and 'astobj2 container check'.
1462 * Added specific CLI commands for bridge inspection. This includes 'bridge
1463 show all', which lists all bridges in the system, and 'bridge show {id}',
1464 which provides specific information about a bridge.
1466 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
1467 ejecting the channels currently in the bridge. If the channels cannot
1468 continue in the dialplan or application that put them in the bridge, they
1471 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1473 * Added commands to inspect and manipulate the registered bridge technologies.
1474 This include 'bridge technology show', which lists the registered bridge
1475 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1476 which controls whether or not a registered bridge technology can be used
1477 during smart bridge operations. If a technology is suspended, it will not
1478 be used when a bridge technology is picked for channels; when unsuspended,
1479 it can be used again.
1481 * The command 'config show help {module} {type} {option}' will show
1482 configuration documentation for modules with XML configuration
1483 documentation. When {module}, {type}, and {option} are omitted, a listing
1484 of all modules with registered documentation is displayed. When {module}
1485 is specified, a listing of all configuration types for that module is
1486 displayed, along with their synopsis. When {module} and {type} are
1487 specified, a listing of all configuration options for that type are
1488 displayed along with their synopsis. When {module}, {type}, and {option}
1489 are specified, detailed information for that configuration option is
1492 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1493 a listing of all installed media sounds available on the system and
1494 detailed information about a sound, respectively.
1496 * 'xmldoc dump' has been added. This CLI command will dump the XML
1497 documentation DOM as a string to the specified file. The Asterisk core
1498 will populate certain XML elements pulled from the source files with
1499 additional run-time information; this command lets a user produce the
1500 XML documentation with all information.
1504 * Parking has been pulled from core and placed into a separate module called
1505 res_parking. See Parking changes below for more details. Configuration for
1506 parking should now be performed in res_parking.conf. Configuration for
1507 parking in features.conf is now unsupported.
1509 * Core attended transfers now have several new options. While performing an
1510 attended transfer, the transferer now has the following options:
1511 - *1 - cancel the attended transfer (configurable via atxferabort)
1512 - *2 - complete the attended transfer, dropping out of the call
1513 (configurable via atxfercomplete)
1514 - *3 - complete the attended transfer, but stay in the call. This will turn
1515 the call into a multi-party bridge (configurable via atxferthreeway)
1516 - *4 - swap to the other party. Once an attended transfer has begun, this
1517 options may be used multiple times (configurable via atxferswap)
1519 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1520 must be on the channel initiating the transfer to have any effect.
1522 * The BRIDGE_FEATURES channel variable would previously only set features for
1523 the calling party and would set this feature regardless of whether the
1524 feature was in caps or in lowercase. Use of a caps feature for a letter
1525 will now apply the feature to the calling party while use of a lowercase
1526 letter will apply that feature to the called party.
1528 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1530 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1531 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1532 activated the dynamic feature.
1534 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1535 only on the channel executing the dynamic feature. Executing a dynamic
1536 feature on the bridge peer in a multi-party bridge will execute it on all
1537 peers of the activating channel.
1539 * You can now have the settings for a channel updated using the FEATURE()
1540 and FEATUREMAP() functions inherited to child channels by setting
1541 FEATURE(inherit)=yes.
1543 * automixmon now supports additional channel variables from automon including:
1544 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1545 and TOUCH_MIXMONITOR_MESSAGE_STOP
1547 * A new general features.conf option 'recordingfailsound' has been added which
1548 allowssetting a failure sound for a user tries to invoke a recording feature
1549 such as automon or automixmon and it fails.
1551 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1552 features.c for atxferdropcall=no to work properly. This option now just
1557 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1558 be used. Given that this can cause the Asterisk log files to grow quickly,
1559 this option should only be used if an external mechanism for log management
1564 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1565 will store the path information for that peer when it registers. Realtime
1566 tables can also use the 'supportpath' field to enable Path header support.
1568 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1569 objectIdentifier. This maps to the supportpath option in sip.conf.
1573 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1574 provides modules a useful abstraction on top of the many storage mechanisms
1575 in Asterisk, including the Asterisk Database, static configuration files,
1576 static Realtime, and dynamic Realtime. It also provides a caching service.
1577 Users can configure a hierarchy of data storage layers for specific modules
1580 * All future modules which utilize Sorcery for object persistence must have a
1581 column named "id" within their schema when using the Sorcery realtime module.
1582 This column must be able to contain a string of up to 128 characters in length.
1584 Security Events Framework
1586 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1587 the "seconds-microseconds" format that it was using previously.
1591 * The Stasis message bus is a publish/subscribe message bus internal to
1592 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1593 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1594 Stasis can be configured in stasis.conf. Note that these parameters operate
1595 at a very low level in Asterisk, and generally will not require changes.
1599 * When a channel driver is configured to enable jiterbuffers, they are now
1600 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1601 is already set for that channel when it enters, such as by the JITTERBUFFER
1602 function, then the existing jitterbuffer will be used and the one set by
1603 the channel driver will not be applied.
1607 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1608 dialplan applications provided by the app_agent_pool module. Agents are
1609 connected with callers using the new AgentRequest dialplan application.
1610 The Agents:<agent-id> device state is available to monitor the status of an
1611 agent. See agents.conf.sample for valid configuration options.
1613 * The updatecdr option has been removed. Altering the names of channels on a
1614 CDR is not supported - the name of the channel is the name of the channel,
1615 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1616 has also been removed, for the same reason.
1618 * The endcall and enddtmf configuration options are removed. Use the
1619 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1620 channel before calling AgentLogin.
1624 * chan_bridge has been removed. Its functionality has been incorporated
1625 directly into the ConfBridge application itself.
1629 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1630 of the specified span and its B-channels. Note that this command should
1631 only be used if you understand the risks it entails.
1633 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1634 A range of channels can be specified to be destroyed. Note that this command
1635 should only be used if you understand the risks it entails.
1637 * Added the CLI command 'dahdi create channels'. A range of channels can be
1638 specified to be created, or the keyword 'new' can be used to add channels
1641 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1642 the exact configured mailbox name. For app_voicemail mailboxes this is
1645 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1649 * IPv6 support has been added. We are now able to bind to and
1650 communicate using IPv6 addresses.
1654 * The /b option has been removed.
1656 * chan_local moved into the system core and is no longer a loadable module.
1660 * Added general support for busy detection.
1662 * Added ECAM command support for Sony Ericsson phones.
1666 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1667 SIP stack. A collection of resource modules provides the bulk of the SIP
1668 functionality. For more information on the new SIP channel driver, see
1669 https://wiki.asterisk.org/wiki/x/JYGLAQ
1673 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1674 using the 'supportpath' setting, either on a global basis or on a peer basis.
1675 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1676 set of proxies by using a pre-loaded route-set defined by the Path headers in
1677 the REGISTER request. See Realtime updates for more configuration information.
1679 * The SIP_CODEC family of variables may now specify more than one codec. Each
1680 codec must be separated by a comma. The first codec specified is the
1681 preferred codec for the offer. This allows a dialplan writer to specify both
1682 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1684 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1685 in the core, and can be filtered out using the 'eventfilter' parameter
1688 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1689 codecs configured for a peer instead of the requested codec.
1691 * The option "register_retry_403" has been added to chan_sip to work around
1692 servers that are known to erroneously send 403 in response to valid
1693 REGISTER requests and allows Asterisk to continue attepmting to connect.
1697 * Added the 'immeddialkey' parameter. If set, when the user presses the
1698 configured key the already entered number will be immediately dialed. This
1699 is useful when the dialplan allows for variable length pattern matching.
1700 Valid options are '*' and '#'.
1702 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1703 milliseconds) before a call forward is considered to not be answered.
1705 * The 'serviceurl' parameter allows Service URLs to be attached to line
1714 * The password option has been disabled, as the AgentLogin application no
1715 longer provides authentication.
1719 * Due to changes in the Asterisk core, this function is no longer needed to
1720 preserve a MixMonitor on a channel during transfer operations and dialplan
1721 execution. It is effectively obsolete.
1725 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1726 deprecated. Use the CHANNEL function instead to access these attributes.
1728 * The 'l' option has been removed. When reading a CDR attribute, the most
1729 recent record is always used. When writing a CDR attribute, all non-finalized
1732 * The 'r' option has been removed, for the same reason as the 'l' option.
1734 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1739 * A new function CDR_PROP has been added. This function lets you set properties
1740 on a channel's active CDRs. This function is write-only. Properties accept
1741 boolean values to set/clear them on the channel's CDRs. Valid properties
1743 - 'party_a' - make this channel the preferred Party A in any CDR between two
1744 channels. If two channels have this property set, the creation time of the
1745 channel is used to determine who is Party A. Note that dialed channels are
1746 never Party A in a CDR.
1747 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1748 application when set to True, and analogous to the 'e' option in ResetCDR
1753 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1754 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1755 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1758 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1759 string, i.e., [[context],extension],priority. If set on a channel, if a
1760 channel leaves a bridge but is not hung up it will resume dialplan execution
1765 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1766 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1767 The value of this setting is ignored when disabled is used for the argument.
1771 * A new function provided by chan_pjsip, this function can be used in
1772 conjunction with the Dial application to construct a dial string that will
1773 dial all contacts on an Address of Record associated with a chan_pjsip
1778 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1779 outbound channel prior to dialing.
1783 * Redirecting reasons can now be set to arbitrary strings. This means
1784 that the REDIRECTING dialplan function can be used to set the redirecting
1785 reason to any string. It also allows for custom strings to be read as the
1786 redirecting reason from SIP Diversion headers.
1790 * The SPEECH_ENGINE function now supports read operations. When read from, it
1791 will return the current value of the requested attribute.
1795 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1796 system as mailbox@context. The rest of the system cannot add @default
1797 to mailbox identifiers for app_voicemail that do not specify a context
1798 any longer. It is a mailbox identifier format that should only be
1799 interpreted by app_voicemail.
1805 res_agi (Asterisk Gateway Interface)
1807 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1809 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1812 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1813 will start the playback of the audio at the position specified. It will
1814 also return the final position of the file in 'endpos'.
1816 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1817 channel variable if the user stopped the file playback or if a remote
1818 entity stopped the playback. If neither stopped the playback, it will
1819 indicate the overall success/failure of the playback. If stopped early,
1820 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1823 * The SAY ALPHA command now accepts an additional parameter to control
1824 whether it specifies the case of uppercase, lowercase, or all letters to
1825 provide functionality similar to SayAlphaCase.
1827 res_ari (Asterisk RESTful Interface) (and others)
1829 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1830 control telephony primitives in Asterisk by remote client. This includes
1831 channels, bridges, endpoints, media, and other fundamental concepts. Users
1832 of ARI can develop their own communications applications, controlling
1833 multiple channels using an HTTP RESTful interface and receiving JSON events
1834 about the objects via a WebSocket connection. ARI can be configured in
1835 Asterisk via ari.conf. For more information on ARI, see
1836 https://wiki.asterisk.org/wiki/x/0YCLAQ
1840 * Parking has been extracted from the Asterisk core as a loadable module,
1841 res_parking. Configuration for parking is now provided by res_parking.conf.
1842 Configuration through features.conf is no longer supported.
1844 * res_parking uses the configuration framework. If an invalid configuration is
1845 supplied, res_parking will fail to load or fail to reload. Previously,
1846 invalid configurations would generally be accepted, with certain errors
1847 resulting in individually disabled parking lots.
1849 * Parked calls are now placed in bridges. While this is largely an
1850 architectural change, it does have implications on how channels in a parking
1851 lot are viewed. For example, commands that display channels in bridges will
1852 now also display the channels in a parking lot.
1854 * The order of arguments for the new parking applications have been modified.
1855 Timeout and return context/exten/priority are now implemented as options,
1856 while the name of the parking lot is now the first parameter. See the
1857 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1858 in-depth information as well as syntax.
1860 * Extensions are by default no longer automatically created in the dialplan to
1861 park calls or pickup parked calls. Generation of dialplan extensions can be
1862 enabled using the 'parkext' configuration option.
1864 * ADSI functionality for parking is no longer supported. The 'adsipark'
1865 configuration option has been removed as a result.
1867 * The PARKINGSLOT channel variable has been deprecated in favor of
1868 PARKING_SPACE to match the naming scheme of the new system.
1870 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1871 channel even when the configuration option 'comebactoorigin' is enabled.
1873 * A new CLI command 'parking show' has been added. This allows a user to
1874 inspect the parking lots that are currently in use.
1875 'parking show <parkinglot>' will also show the parked calls in a specific
1878 * The CLI command 'parkedcalls' is now deprecated in favor of
1879 'parking show <parkinglot>'.
1881 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1882 can be used to get a list of parked calls for a specific parking lot.
1884 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1885 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1886 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1887 longer a required argument.
1889 * The ParkAndAnnounce application is now provided through res_parking instead
1890 of through the separate app_parkandannounce module.
1892 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1893 by default. Instead, it will follow the timeout rules of the parking lot. The
1894 old behavior can be reproduced by using the 'c' option.
1896 * Dynamic parking lots will now fail to be created under the following
1898 - if the parking lot specified by PARKINGDYNAMIC does not exist
1899 - if they require exclusive park and parkedcall extensions which overlap
1900 with existing parking lots.
1902 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1903 currently contain no calls. Dynamic parking lots containing parked calls
1904 will persist through the reloads without alteration.
1906 * If 'parkext_exclusive' is set for a parking lot and that extension is
1907 already in use when that parking lot tries to register it, this is now
1908 considered a parking system configuration error. Configurations which do
1909 this will be rejected.
1911 * Added channel variable PARKER_FLAT. This contains the name of the extension
1912 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1913 comebacktoorigin is disabled, but the dialplan or an external control
1914 mechanism wants to use the extension in the park-dial context that was
1915 generated to re-dial the parker on timeout.
1917 res_pjsip (and many others)
1919 * A large number of resource modules make up the SIP stack based on pjsip.
1920 The chan_pjsip channel driver users these resource modules to provide
1921 various SIP functionality in Asterisk. The majority of configuration for
1922 these modules is performed in pjsip.conf. Other modules may use their
1923 own configuration files.
1925 * Added 'set_var' option for an endpoint. For each variable specified that
1926 variable gets set upon creation of a channel involving the endpoint.
1930 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1931 them, an Asterisk-specific version of PJSIP needs to be installed.
1932 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1934 res_statsd/res_chan_stats
1936 * A new resource module, res_statsd, has been added, which acts as a statsd
1937 client. This module allows Asterisk to publish statistics to a statsd
1938 server. In conjunction with res_chan_stats, it will publish statistics about
1939 channels to the statsd server. It can be configured via res_statsd.conf.
1943 * Device state for XMPP buddies is now available using the following format:
1944 XMPP/<client name>/<buddy address>
1945 If any resource is available the device state is considered to be not in use.
1946 If no resources exist or all are unavailable the device state is considered
1953 Realtime/Database Scripts
1955 * Asterisk previously included example db schemas in the contrib/realtime/
1956 directory of the source tree. This has been replaced by a set of database
1957 migrations using the Alembic framework. This allows you to use alembic to
1958 initialize the database for you. It will also serve as a database migration
1959 tool when upgrading Asterisk in the future.
1961 See contrib/ast-db-manage/README.md for more details.
1965 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1966 This python script will convert an existing sip.conf file to a
1967 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1968 is meant to be an aid in converting an existing chan_sip configuration to
1969 a chan_pjsip configuration, but it is expected that configuration beyond
1970 what the script provides will be needed.
1972 ------------------------------------------------------------------------------
1973 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1974 ------------------------------------------------------------------------------
1978 * The Asterisk build system will now build and install a shared library
1979 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1980 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1981 that Asterisk can ensure that these functions do *not* get called by any
1982 modules that are loaded into Asterisk, since they should only be called once
1983 in any single process. If desired, this feature can be disabled by supplying
1984 the "--disable-asteriskssl" option to the configure script.
1986 * A new make target, 'full', has been added to the Makefile. This performs
1987 the same compilation actions as make all, but will also scan the entirety of
1988 each source file for documentation. This option is needed to generate AMI
1989 event documentation. Note that your system must have Python in order for
1990 this make target to succeed.
1992 * The optimization portion of the build system has been reworked to avoid
1993 broken builds on certain architectures. All architecture-specific
1994 optimization has been removed in favor of using -march=native to allow gcc
1995 to detect the environment in which it is running when possible. This can
1996 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1998 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1999 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2001 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2002 previously parsed the header file to obtain the version of Asterisk, you
2003 will now have to go through Asterisk to get the version information.
2011 * Added 'F()' option. Similar to the dial option, this can be supplied with
2012 arguments indicating where the callee should go after the caller is hung up,
2013 or without options specified, the priority after the Queue will be used.
2018 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2019 all non-admin participants on a conference. The confbridge configuration
2020 file also allows for the default sounds played to all conference users when
2021 this occurs to be overriden using sound_participants_unmuted and
2022 sound_participants_muted.
2024 * Added menu action participant_count. This will playback the number of
2025 current participants in a conference.
2027 * Added announcement configuration option to user profile. If set the sound
2028 file will be played to the user, and only the user, upon joining the
2031 * Added record_file_append option that defaults to "yes", but if set to no
2032 will create a new file between each start/stop recording.
2037 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2038 channels respectively before the callee channels are called.
2043 * Added support for IPv6.
2045 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2046 external process will cause the current playlist to be cleared, including
2047 stopping any audio file that is currently playing. This is useful when you
2048 want to interrupt audio playback only when specific DTMF is entered by the
2054 * A new option, 'I' has been added to app_followme. By setting this option,
2055 Asterisk will not update the caller with connected line changes when they
2056 occur. This is similar to app_dial and app_queue.
2058 * The 'N' option is now ignored if the call is already answered.
2060 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2061 and caller channels respectively before the callee channels are called.
2063 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2069 * MixMonitor hooks now have IDs associated with them which can be used to
2070 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2071 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2072 now accepts that ID as an argument.
2074 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2075 indicated mailboxes.
2080 * The connect action in app_mysql now allows you to specify a port number to
2081 connect to. This is useful if you run a MySQL server on a non-standard
2087 * Increased the default number of allowed destinations from 5 to 12.
2092 * The app_page application now no longer depends on DAHDI or app_meetme. It
2093 has been re-architected to use app_confbridge internally.
2098 * Added queue options autopausebusy and autopauseunavail for automatically
2099 pausing a queue member when their device reports busy or congestion.
2101 * The 'ignorebusy' option for queue members has been deprecated in favor of
2102 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2103 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2104 per interface basis. Individual ringinuse values can now be set in
2105 queues.conf via an argument to member definitions. Lastly, the queue
2106 'ringinuse' setting now only determines defaults for the per member
2107 'ringinuse' setting and does not override per member settings like it does
2108 in earlier versions.
2110 * Added 'F()' option. Similar to the dial option, this can be supplied with
2111 arguments indicating where the callee should go after the caller is hung up,
2112 or without options specified, the priority after the Queue will be used.
2114 * Added new option log_member_name_as_agent, which will cause the membername to
2115 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2116 state_interface has been set.
2118 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2120 * App_queue will now play periodic announcements for the caller that
2121 holds the first position in the queue while waiting for answer.
2125 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2126 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2127 changed arguments to SayUnixTime so that every option is truly optional even
2128 when using multiple options (so that j option could be used without having to
2129 manually specify timezone and format) There are other benefits, e.g., format
2130 can now be used without specifying time zone as well.
2135 * Addition of the VM_INFO function - see Function changes.
2137 * The imapserver, imapport, and imapflags configuration options can now be
2138 overriden on a user by user basis.
2140 * When voicemail plays a message's envelope with saycid set to yes, when
2141 reaching the caller id field it will play a recording of a file with the same
2142 base name as the sender's callerid if there is a similarly named file in
2143 <astspooldir>/recordings/callerids/
2145 * Voicemails now contains a unique message identifier "msg_id", which is stored
2146 in the message envelope with the sound files. IMAP backends will now store
2147 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2148 backends will store the message identifier in a "msg_id" column. See
2149 UPGRADE.txt for more information.
2151 * Added VoiceMailPlayMsg application. This application will play a single
2152 voicemail message from a mailbox. The result of the application, SUCCESS or
2153 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2158 * Hangup handlers can be attached to channels using the CHANNEL() function.
2159 Hangup handlers will run when the channel is hung up similar to the h
2160 extension. The hangup_handler_push option will push a GoSub compatible
2161 location in the dialplan onto the channel's hangup handler stack. The
2162 hangup_handler_pop option will remove the last added location, and optionally
2163 replace it with a new GoSub compatible location. The hangup_handler_wipe
2164 option will remove all locations on the stack, and optionally add a new
2167 * The expression parser now recognizes the ABS() absolute value function,
2168 which will convert negative floating point values to positive values.
2170 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2171 control of faxdetect.
2173 * Addition of the VM_INFO function that can be used to retrieve voicemail
2174 user information, such as the email address and full name.
2175 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2178 * The REDIRECTING function now supports the redirecting original party id
2181 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2182 lets you set some of the configuration options from the [general] section
2183 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2184 the key sequence used to activate built-in features, such as blindxfer,
2185 and automon. See the built-in documentation for details.
2187 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2188 instead of simply the uri. This is the format that MessageSend() can use
2189 in the from parameter for outgoing SIP messages.
2191 * Added the PRESENCE_STATE function. This allows retrieving presence state
2192 information from any presence state provider. It also allows setting
2193 presence state information from a CustomPresence presence state provider.
2194 See AMI/CLI changes for related commands.
2196 * Added the AMI_CLIENT function to make manager account attributes available
2197 to the dialplan. It currently supports returning the current number of
2198 active sessions for a given account.
2200 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2201 and the REDIRECTING functions.
2209 * Added a manager event "LocalBridge" for local channel call bridges between
2210 the two pseudo-channels created.
2215 * Added dialtone_detect option for analog ports to disconnect incoming
2216 calls when dialtone is detected.
2218 * Added option colp_send to send ISDN connected line information. Allowed
2219 settings are block, to not send any connected line information; connect, to
2220 send connected line information on initial connect; and update, to send
2221 information on any update during a call. Default is update.
2223 * Add options namedcallgroup and namedpickupgroup to support installations
2224 where a higher number of groups (>64) is required.
2226 * Added support to use private party ID information with PRI calls.
2231 * A new channel driver named chan_motif has been added which provides support for
2232 Google Talk and Jingle in a single channel driver. This new channel driver includes
2233 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2234 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2235 specification, current Google Jingle specification, and the original Google Talk
2241 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2242 globally and overriden on a peer by peer basis.
2244 * Direct media functionality has been added. Options in config are:
2245 directmedia (directrtp) and directrtpsetup (earlydirect)
2247 * ChannelUpdate events now contain a CallRef header.
2252 * Asterisk will no longer substitute CID number for CID name in the display
2253 name field if CID number exists without a CID name. This change improves
2254 compatibility with certain device features such as Avaya IP500's directory
2257 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2258 created using that setting to not be removed during SIP reload.
2260 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2261 request with a "Record:" header, this will turn the requested feature on/off.
2262 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2263 dynamic features must be enabled and configured properly on the requesting
2264 channel for this to function properly.
2266 * Add support to realtime for the 'callbackextension' option.
2268 * When multiple peers exist with the same address, but differing
2269 callbackextension options, incoming requests that are matched by address
2270 will be matched to the peer with the matching callbackextension if it is
2273 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2274 which set the force_rport and comedia options automatically if Asterisk
2275 detects that an incoming SIP request crossed a NAT after being sent by
2276 the remote endpoint.
2278 * The default global nat setting in sip.conf has been changed from force_rport
2279 to auto_force_rport.
2281 * NAT settings are now a combinable list of options. The equivalent of the
2282 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2284 * Adds an option send_diversion which can be disabled to prevent
2285 diversion headers from automatically being added to INVITE requests.
2287 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2288 be sent to the remote host at a given interval to keep the NAT mapping open.
2289 This can be enabled using the keepalive configuration option.
2291 * Add option 'tonezone' to specify country code for indications. This option
2292 can be set both globally and overridden for specific peers.
2294 * The SIP Security Events Framework now supports IPv6.
2296 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2297 between multiple user agents. When set, for directmedia reinvites,
2298 Asterisk will not send an immediate reinvite on an incoming call leg. This
2299 option is useful when peered with another SIP user agent that is known to
2300 send immediate direct media reinvites upon call establishment.
2302 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2305 * Add options subminexpiry and submaxexpiry to set limits of subscription
2306 timer independently from registration timer settings. The setting of the
2307 registration timer limits still is done by options minexpiry, maxexpiry
2308 and defaultexpiry. For backwards compatibility the setting of minexpiry
2309 and maxexpiry also is used to configure the subscription timer limits if
2310 subminexpiry and submaxexpiry are not set in sip.conf.
2312 * Set registration timer limits to default values when reloading sip
2313 configuration and values are not set by configuration.
2315 * Add options namedcallgroup and namedpickupgroup to support installations
2316 where a higher number of groups (>64) is required.
2318 * When a MESSAGE request is received, the address the request was received from
2319 is now saved in the SIP_RECVADDR variable.
2321 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2322 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2323 the ANI2/OLI information is set on the channel, which can be retrieved using
2324 the CALLERID function.
2326 * Peers can now be configured to support negotiation of ICE candidates using
2327 the setting icesupport. See res_rtp_asterisk changes for more information.
2329 * Added support for format attribute negotiation. See the Codecs changes for
2332 * Extra headers specified with SIPAddHeader are sent with the REFER message
2333 when using Transfer application. See refer_addheaders in sip.conf.sample.
2335 * Added support to use private party ID information with calls.
2337 * Adds an option discard_remote_hold_retrieval that when set stops telling
2338 the peer to start music on hold.
2343 * Added skinny version 17 protocol support.
2347 --------------------
2348 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2350 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2351 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2352 as per the UNISTIM protocol.
2354 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2355 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2357 * Added ability to use multiple lines for a single phone. This allows multiple
2358 calls to occur on a single phone, using callwaiting and switching between calls.
2360 * Added option 'sharpdial' allowing end dialing by pressing # key
2362 * Added option 'interdigit_timer' to control phone dial timeout
2364 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2366 * Added global 'debug' option, that enables debug in channel driver
2368 * Added ability to translate on-screen menu in multiple languages. Tested on
2369 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2370 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2373 * In addition to English added French and Russian languages for on-screen menus
2375 * Reworked dialing number input: added dialing by timeout, immediate dial on
2376 on dialplan compare, phone number length now not limited by screen size
2378 * Added ability to pickup a call using features.conf defined value and
2384 * Add options namedcallgroup and namedpickupgroup to support installations
2385 where a higher number of groups (>64) is required.
2387 * Added support to use private party ID information with calls.
2392 * The minimum DTMF duration can now be configured in asterisk.conf
2393 as "mindtmfduration". The default value is (as before) set to 80 ms.
2394 (previously it was only available in source code)
2396 * Named ACLs can now be specified in acl.conf and used in configurations that
2397 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2398 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2399 working ACL. In addition, some CLI commands have been added to provide
2400 show information and allow for module reloading - see CLI Changes.
2402 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2403 items (separated by commas), and items in the rule can be negated by prefixing
2404 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2405 longer necessray to control the order that the 'permit' and 'deny' columns are
2406 returned from queries.
2408 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2409 be used within the dynamic weight attribute when specifying a mapping.
2411 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2412 header, instead of putting the user defined event name there. When enabled
2413 the UserDefType header is added for user defined events. This feature is
2414 enabled with the setting show_user_defined.
2416 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2417 line purposes use the following variables instead of their macro equivalents:
2418 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2419 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
2420 cc_callback_macro in channel configurations.
2422 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
2425 * Call files now support the "early_media" option to connect with an outgoing
2426 extension when early media is received.
2428 * Added support to use private party ID information with calls.
2433 * A new channel variable, AGIEXITONHANGUP, has been added which allows
2434 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
2435 AGI application would exit immediately after a channel hangup is detected.
2437 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
2438 are resolved and each address is attempted in turn until one succeeds or
2442 AMI (Asterisk Manager Interface)
2444 * The originate action now has an option "EarlyMedia" that enables the
2445 call to bridge when we get early media in the call. Previously,
2446 early media was disregarded always when originating calls using AMI.
2448 * Added setvar= option to manager accounts (much like sip.conf)
2450 * Originate now generates an error response if the extension given is not found
2453 * MixMonitor will now show IDs associated with the mixmonitor upon creating
2454 them if the i(variable) option is used. StopMixMonitor will accept
2455 MixMonitorID as an option to close specific MixMonitors.
2457 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
2458 updated to include information about peers configured with
2459 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
2460 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
2461 returned if auto_force_rport is not enabled.
2463 * Added SIPpeerstatus manager command which will generate PeerStatus events
2464 similar to the existing PeerStatus events found in chan_sip on demand.
2466 * Hangup now can take a regular expression as the Channel option. If you want
2467 to hangup multiple channels, use /regex/ as the Channel option. Existing
2468 behavior to hanging up a single channel is unchanged, but if you pass a regex,
2469 the manager will send you a list of channels back that were hung up.
2471 * Support for IPv6 addresses has been added.
2473 * AMI Events can now be documented in the Asterisk source. Note that AMI event
2474 documentation is only generated when Asterisk is compiled using 'make full'.
2475 See the CLI section for commands to display AMI event information.
2477 * The AMI Hangup event now includes the AccountCode header so you can easily
2478 correlate with AMI Newchannel events.
2480 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2481 the StateInterface of the queue member.
2483 * Added AMI event SessionTimeout in the Call category that is issued when a
2484 call is terminated due to either RTP stream inactivity or SIP session timer
2487 * CEL events can now contain a user defined header UserDefType. See core
2488 changes for more information.
2490 * OOH323 ChannelUpdate events now contain a CallRef header.
2492 * Added PresenceState command. This command will report the presence state for
2493 the given presence provider.
2495 * Added Parkinglots command. This will list all parking lots as a series of
2496 AMI Parkinglot events.
2498 * Added MessageSend command. This behaves in the same manner as the
2499 MessageSend application, and is a technolgoy agnostic mechanism to send out
2500 of call text messages.
2502 * Added "message" class authorization. This grants an account permission to
2503 send out of call messages. Write-only.
2508 * The "dialplan add include" command has been modified to create context a context
2509 if one does not already exist. For instance, "dialplan add include foo into bar"
2510 will create context "bar" if it does not already exist.
2512 * A "dialplan remove context" command has been added to remove a context from
2515 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2516 filenames of all running mixmonitors on a channel.
2518 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2519 numeric instead of 0, 1, or 2.
2521 * "stun show status" will show a table describing how the STUN client is
2524 * "acl show [named acl]" will show information regarding a Named ACL. The
2525 acl module can be reloaded with "reload acl".
2527 * Added CLI command to display AMI event information - "manager show events",
2528 which shows a list of all known and documented AMI events, and "manager show
2529 event [event name]", which shows detail information about a specific AMI
2532 * The result of the CLI command "queue show" now includes the state interface
2533 information of the queue member.
2535 * The command "core set verbose" will now set a separate level of logging for
2536 each remote console without affecting any other console.
2538 * Added command "cdr show pgsql status" to check connection status
2540 * "sip show channel" will now display the complete route set.
2542 * Added "presencestate list" command. This command will list all custom
2543 presence states that have been set by using the PRESENCE_STATE dialplan
2546 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2547 command. This changes a custom presence to a new state.
2552 * Codec lists may now be modified by the '!' character, to allow succinct
2553 specification of a list of codecs allowed and disallowed, without the
2554 requirement to use two different keywords. For example, to specify all
2555 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2557 * Add support for parsing SDP attributes, generating SDP attributes, and
2558 passing it through. This support includes codecs such as H.263, H.264, SILK,
2559 and CELT. You are able to set up a call and have attribute information pass.
2560 This should help considerably with video calls.
2562 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2563 just like the GSM codec.
2567 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2568 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2572 * Asterisk version and build information is now logged at the beginning of a
2575 * Threads belonging to a particular call are now linked with callids which get
2576 added to any log messages produced by those threads. Log messages can now be
2577 easily identified as involved with a certain call by looking at their call id.
2578 Call ids may also be attached to log messages for just about any case where
2579 it can be determined to be related to a particular call.
2581 * Each logging destination and console now have an independent notion of the
2582 current verbosity level. Logger.conf now allows an optional argument to
2583 the 'verbose' specifier, indicating the level of verbosity sent to that
2584 particular logging destination. Additionally, remote consoles now each
2585 have their own verbosity level. The command 'core set verbose' will now set
2586 a separate level for each remote console without affecting any other
2592 * Added 'announcement' option which will play at the start of MOH and between
2593 songs in modes of MOH that can detect transitions between songs (eg.
2599 * New per parking lot options: comebackcontext and comebackdialtime. See
2600 configs/features.conf.sample for more details.
2602 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2605 * Channel variable PARKEDCALL is now set with the name of the parking lot
2606 when a timeout occurs.
2612 CDR Postgresql Driver
2614 * Added command "cdr show pgsql status" to check connection status
2617 CDR Adaptive ODBC Driver
2619 * Added schema option for databases that support specifying a schema.
2627 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2628 CALENDAR_WRITE has completed successfully.
2633 * A new option, 'probation' has been added to rtp.conf
2634 RTP in strictrtp mode can now require more than 1 packet to exit learning
2635 mode with a new source (and by default requires 4). The probation option
2636 allows the user to change the required number of packets in sequence to any
2637 desired value. Use a value of 1 to essentially restore the old behavior.
2638 Also, with strictrtp on, Asterisk will now drop all packets until learning
2639 mode has successfully exited. These changes are based on how pjmedia handles
2640 media sources and source changes.
2642 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2643 enabled or disabled using the icesupport setting. A variety of other
2644 settings have been introduced to configure STUN/TURN connections.
2649 * A new module, res_corosync, has been introduced. This module uses the
2650 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2651 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2652 Device State (presence) information. This module is very similar to, and
2653 is a replacement for the res_ais module that was in previous releases of
2659 * This module adds a cleaned up, drop-in replacement for res_jabber called
2660 res_xmpp. This provides the same externally facing functionality but is
2661 implemented differently internally. res_jabber has been deprecated in favor
2662 of res_xmpp; please see the UPGRADE.txt file for more information.
2667 * The safe_asterisk script has been updated to allow several of its parameters
2668 to be set from environment variables. This also enables a custom run
2669 directory of Asterisk to be specified, instead of defaulting to /tmp.
2671 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2672 its value to determine the directory to assume is the top-level directory of
2673 the source tree. If the variable is not set, it defaults to the current
2674 behavior and uses the current working directory.
2676 ------------------------------------------------------------------------------
2677 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2678 ------------------------------------------------------------------------------
2682 * Asterisk now has protocol independent support for processing text messages
2683 outside of a call. Messages are routed through the Asterisk dialplan.
2684 SIP MESSAGE and XMPP are currently supported. There are options in
2685 jabber.conf and sip.conf to allow enabling these features.
2686 -> jabber.conf: see the "sendtodialplan" and "context" options.
2687 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2688 and "outofcall_message_context" options.
2689 The MESSAGE() dialplan function and MessageSend() application have been
2690 added to go along with this functionality. More detailed usage information
2691 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2692 * If real-time text support (T.140) is negotiated, it will be preferred for
2693 sending text via the SendText application. For example, via SIP, messages
2694 that were once sent via the SIP MESSAGE request would be sent via RTP if
2695 T.140 text is negotiated for a call.
2699 * parkedmusicclass can now be set for non-default parking lots.
2701 Asterisk Manager Interface
2702 --------------------------
2703 * PeerStatus now includes Address and Port.
2704 * Added Hold events for when the remote party puts the call on and off hold
2705 for chan_dahdi ISDN channels.
2706 * Added new action MeetmeListRooms to list active conferences (shows same
2707 data as "meetme list" at the CLI).
2708 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2709 Description field that is set by 'description' in the channel configuration
2711 * Added Uniqueid header to UserEvent.
2712 * Added new action FilterAdd to control event filters for the current session.
2713 This requires the system permission and uses the same filter syntax as
2714 filters that can be defined in manager.conf
2715 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2716 versions had some instances of the event converted, but others were left
2717 as-is. All Unlink events should now be converted to Bridge events. The AMI
2718 protocol version number was incremented to 1.2 as a result of this change.
2720 Asterisk HTTP Server
2721 --------------------------
2722 * The HTTP Server can bind to IPv6 addresses.
2725 --------------------------
2726 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2727 with busydetect. usage example: busypattern=200,200,200,600
2730 --------------------------
2731 * New 'gtalk show settings' command showing the current settings loaded from
2733 * The 'logger reload' command now supports an optional argument, specifying an
2734 alternate configuration file to use.
2735 * 'dialplan add extension' command will now automatically create a context if
2736 the specified context does not exist with a message indicated it did so.
2737 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2738 Description field which can be populated with 'description' in the channel
2739 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2742 --------------------------
2743 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2744 thus allowing records which do NOT match the specified filter.
2745 * Added ability to log CONGESTION calls to CDR
2748 --------------------------
2749 * Ability to define custom SILK formats in codecs.conf.
2750 * Addition of speex32 audio format with translation.
2751 * CELT codec pass-through support and ability to define
2752 custom CELT formats in codecs.conf.
2753 * Ability to read raw signed linear files with sample rates
2754 ranging from 8khz - 192khz. The new file extensions introduced
2755 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2756 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2757 Skinny, H.323, etc) can still only support the following codecs:
2758 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2759 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2760 Video: h261, h263, h263p, h264, mpeg4
2765 --------------------------
2766 * New highly optimized and customizable ConfBridge application capable of
2767 mixing audio at sample rates ranging from 8khz-96khz.
2768 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2769 and bridge profiles on a channel.
2770 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2771 about a conference such as locked status and number of parties, admins,
2773 * Addition of video_mode option in confbridge.conf for adding video support
2774 into a bridge profile.
2775 * Addition of the follow_talker video_mode in confbridge.conf. This video
2776 mode dynamically switches the video feed to always display the loudest talker
2777 supplying video in the conference.
2781 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2782 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2783 variables from asterisk.conf.
2787 * Addition of the JITTERBUFFER dialplan function. This function allows
2788 for jitterbuffering to occur on the read side of a channel. By using
2789 this function conference applications such as ConfBridge and MeetMe can
2790 have the rx streams jitterbuffered before conference mixing occurs.
2791 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2793 * Added STRREPLACE function. This function let's the user search a variable
2794 for a given string to replace with another string as many times as the
2795 user specifies or just throughout the whole string.
2796 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2797 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2798 * Added extensions to chan_ooh323 in function CHANNEL()
2800 libpri channel driver (chan_dahdi) DAHDI changes
2801 --------------------------
2802 * Added moh_signaling option to specify what to do when the channel's bridged
2803 peer puts the ISDN channel on hold.
2804 * Added display_send and display_receive options to control how the display ie
2805 is handled. To send display text from the dialplan use the SendText()
2806 application when the option is enabled.
2807 * Added mcid_send option to allow sending a MCID request on a span.
2810 --------------------------
2811 * Added setvar option to calendar.conf to allow setting channel variables on
2812 notification channels.
2813 * Added "calendar show types" CLI command to list registered calendar
2817 --------------------------
2818 * Added two new options, r and t with file name arguments to record
2819 single direction (unmixed) audio recording separate from the bidirectional
2820 (mixed) recording. The mixed file name argument is optional now as long
2821 as at least one recording option is used.
2824 --------------------------
2825 * Added a new option, l, which will disable local call optimization for
2826 channels involved with the FollowMe thread. Use this option to improve
2827 compatability for a FollowMe call with certain dialplan apps, options, and
2831 --------------------------
2832 * Added option "k" that will automatically close the conference when there's
2833 only one person left when a user exits the conference.
2836 --------------------------
2837 * cel_pgsql now supports the 'extra' column for data added using the
2838 CELGenUserEvent() application.
2841 --------------------------
2842 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2843 in the sample extensions.lua file for syntax details.
2844 * Applications that perform jumps in the dialplan such as Goto will now
2845 execute properly. When pbx_lua detects that the context, extension, or
2846 priority we are executing on has changed it will immediately return control
2847 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2848 the priority after the currently executing priority.
2849 * An autoservice is now started by default for pbx_lua channels. It can be
2850 stopped and restarted using the autoservice_stop() and autoservice_start()
2854 --------------------------
2855 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2856 into a FAXStatus event with an 'Operation' header that will be either
2857 'send', 'receive', and 'gateway'.
2858 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2859 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2860 feature will handle converting a fax call between an audio T.30 fax terminal
2861 and an IFP T.38 fax terminal.
2865 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2866 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2867 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2871 * Added general option negative_penalty_invalid default off. when set
2872 members are seen as invalid/logged out when there penalty is negative.
2873 for realtime members when set remove from queue will set penalty to -1.
2874 * Added queue option autopausedelay when autopause is enabled it will be
2875 delayed for this number of seconds since last successful call if there
2876 was no prior call the agent will be autopaused immediately.
2877 * Added member option ignorebusy this when set and ringinuse is not
2878 will allow per member control of multiple calls as ringinuse does for
2883 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2885 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2886 one participant left (much like a normal call bridge)
2887 * Added extra argument to Originate to set timeout.
2891 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2892 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2893 utility in the UTILS section of menuselect. If an existing astdb is found and no
2894 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2895 convert an existing astdb to the SQLite3 version automatically at runtime.
2899 * Modules marked as deprecated are no longer marked as building by default. Enabling
2900 these modules is still available via menuselect.
2904 * authdebug is now disabled by default. To enable this functionaility again
2905 set authdebug = yes in iax.conf.
2909 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2910 releases it was disabled.
2914 * The PBX core previously made a call with a non-existing extension test for
2915 extension s@default and jump there if the extension existed.
2916 This was a bad default behaviour and violated the principle of least surprise.
2917 It has therefore been changed in this release. It may affect some
2918 applications and configurations that rely on this behaviour. Most channel
2919 drivers have avoided this for many releases by testing whether the extension
2920 called exists before starting the PBX and generating a local error.
2921 This behaviour still exists and works as before.
2923 Extension "s" is used when no extension is given in a channel driver,
2924 like immediate answer in DAHDI or calling to a domain with no user part
2927 ------------------------------------------------------------------------------
2928 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2929 ------------------------------------------------------------------------------
2933 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2934 now defaults to force_rport. It is very important that phones requiring nat=no be
2935 specifically set as such instead of relying on the default setting. If at all
2936 possible, all devices should have nat settings configured in the general section as
2937 opposed to configuring nat per-device.
2938 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2939 codecs sent in response to an INVITE to the single most preferred codec.
2940 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2941 to be used for the outgoing call. It must be one of the codecs configured
2943 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2944 to be used for holding a private key. If tlsprivatekey is not specified,
2945 tlscertfile is searched for both public and private key.
2946 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2947 outbound client connections to be specified.
2948 * The sendrpid parameter has been expanded to include the options
2949 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2950 header to be sent (equivalent to setting sendrpid=yes) and setting
2951 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2952 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2953 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2954 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2955 will accept the SDP even if the SDP version number is not properly incremented,
2956 but will generate a warning in the log indicating that the SIP peer that sent
2957 the SDP should have the 'ignoresdpversion' option set.
2958 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2959 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2960 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2961 remote side requests it and disables symmetric RTP support. Setting it to
2962 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2963 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2964 and enables symmetric RTP support.
2965 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2966 response. This permits the master channel to know how each channel dialled
2967 in a multi-channel setup resolved in an individual way. This carries a
2968 performance penalty and can be disabled in sip.conf using the
2969 'storesipcause' option.
2970 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2971 configuration for the externip and externhost options when tcp or tls is used.
2972 * Added support for message body (stored in content variable) to SIP NOTIFY message
2973 accessible via AMI and CLI.
2974 * Added 'media_address' configuration option which can be used to explicitly specify
2975 the IP address to use in the SDP for media (audio, video, and text) streams.
2976 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2977 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2979 * Added 'use_q850_reason' configuration option for generating and parsing
2980 if available Reason: Q.850;cause=<cause code> header. It is implemented
2981 in some gateways for better passing PRI/SS7 cause codes via SIP.
2982 * When dialing SIP peers, a new component may be added to the end of the dialstring
2983 to indicate that a specific remote IP address or host should be used when dialing
2984 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2985 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2986 ability to selectively force bridged channels to also be encrypted is also
2987 implemented. Branching in the dialplan can be done based on whether or not
2988 a channel has secure media and/or signaling.
2989 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2991 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2992 Charge messages to snom phones.
2993 * Added support for G.719 media streams.
2994 * Added support for 16khz signed linear media streams.
2995 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2996 RTP has been outfitted with the same abilities.
2997 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2998 available in device configurations as well as in the dial plan.
2999 * Addition of the 'subscribe_network_change' option for turning on and off
3000 res_stun_monitor module support in chan_sip.
3001 * Addition of the 'auth_options_requests' option for turning on and off
3002 authentication for OPTIONS requests in chan_sip.
3006 * Add #tryinclude statement for config files. This provides the same
3007 functionality as the #include statement however an asterisk module will
3008 still load if the filename does not exist. Using the #include statement
3009 Asterisk will not allow the module to load.
3013 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3014 on realtime updates.
3015 * Added the ability for chan_iax2 to inform the dialplan whether or not
3016 encryption is being used. This interoperates with the SIP SRTP implementation
3017 so that a secure SIP call can be bridged to a secure IAX call when the
3018 dialplan requires bridged channels to be "secure".
3019 * Addition of the 'subscribe_network_change' option for turning on and off
3020 res_stun_monitor module support in chan_iax.
3025 * Added ability to preset channel variables on indicated lines with the setvar
3026 configuration option. Also, clearvars=all resets the list of variables back
3028 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3029 See configs/res_pktccops.conf for more information.
3031 XMPP Google Talk/Jingle changes
3032 -------------------------------
3033 * Added the externip option to gtalk.conf.
3034 * Added the stunaddr option to gtalk.conf which allows for the automatic
3035 retrieval of the external ip from a stun server.
3039 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3040 match to a partial channel name.
3041 * Added .m3u support for Mp3Player application.
3042 * Added progress option to the app_dial D() option. When progress DTMF is
3043 present, those values are sent immediately upon receiving a PROGRESS message
3044 regardless if the call has been answered or not.
3045 * Added functionality to the app_dial F() option to continue with execution
3046 at the current location when no parameters are provided.
3047 * Added the 'a' option to app_dial to answer the calling channel before any
3048 announcements or macros are executed.
3049 * Modified app_dial to set answertime when the called channel answers even if
3050 the called channel hangs up during playback of an announcement.
3051 * Modified app_dial 'r' option to support an additional parameter to play an
3052 indication tone from indications.conf
3053 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3054 to cycle through the next available channel. By default this is still '*'.
3055 * Added x() option to app_chanspy. This option allows DTMF to be set to
3056 exit the application.
3057 * The Voicemail application has been improved to automatically ignore messages
3058 that only contain silence.
3059 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3060 associated mailbox(es) to be greetings-only.
3061 * The ChanSpy application now has the 'S' option, which makes the application
3062 automatically exit once it hits a point where no more channels are available
3064 * The ChanSpy application also now has the 'E' option, which spies on a single
3065 channel and exits when that channel hangs up.
3066 * The MeetMe application now turns on the DENOISE() function by default, for
3067 each participant. In our tests, this has significantly decreased background
3068 noise (especially noisy data centers).
3069 * Voicemail now permits storage of secrets in a separate file, located in the
3070 spool directory of each individual user. The control for this is located in
3071 the "passwordlocation" option in voicemail.conf. Please see the sample
3072 configuration for more information.
3073 * The ChanIsAvail application now exposes the returned cause code using a separate
3074 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3075 * Added 'd' option to app_followme. This option disables the "Please hold"
3077 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3078 received will terminate recording.
3079 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3080 Previously the folder could only be set per context, but has now been extended
3081 using the imapfolder option.
3082 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3083 * Voicemail now allows the pager date format to be specified separately from the
3085 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3086 to allow joining, leaving, and sending text to group chats.
3087 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3088 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3089 to all paged phones (and optionally excluding the caller's one using the new
3090 option 'n') before the call is bridged.
3091 * The 'f' option to Dial has been augmented to take an optional argument. If no
3092 argument is provided, the 'f' option works as it always has. If an argument is
3093 provided, then the connected party information of all outgoing channels created
3094 during the Dial will be set to the argument passed to the 'f' option.
3095 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3097 * The OSP lookup application adds in/outbound network ID, optional security,
3098 number portability, QoS reporting, destination IP port, custom info and service
3100 * Added new application VMSayName that will play the recorded name of the voicemail
3101 user if it exists, otherwise will play the mailbox number.
3102 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3103 retrieve state for a particular bridge, where <name> is the conference name
3104 * app_directory now allows exiting at any time using the operator or pound key.
3105 * Voicemail now supports setting a locale per-mailbox.
3106 * Two new applications are provided for declining counting phrases in multiple
3107 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3109 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3111 * Voicemail now includes rdnis within msgXXXX.txt file.
3112 * ExternalIVR now supports IPv6 addresses.
3113 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3114 at https://wiki.asterisk.org/wiki/x/oQBB
3115 * ParkedCall and Park can now specify the parking lot to use.
3119 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3120 over SRV records associated with a specific service. From the CLI, type
3121 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3122 details on how these may be used.
3123 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3124 pitch of a channel's tx and rx audio streams.
3125 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3126 setting various connected line and redirecting party information.
3127 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3128 support ISDN subaddressing.
3129 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3130 * For DAHDI channels, the CHANNEL() dialplan function now allows
3131 the dialplan to request changes in the configuration of the active
3132 echo canceller on the channel (if any), for the current call only.
3135 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3137 The possible values are:
3139 on - normal mode (the echo canceller is actually reinitialized)
3141 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3143 voice - voice mode (returns from FAX mode, reverting the changes that
3144 were made when FAX mode was requested)
3145 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3146 and setting variables on the channel which created the current channel.
3147 Administrators should take care to avoid naming conflicts, when multiple
3148 channels are dialled at once, especially when used with the Local channel
3149 construct (which all could set variables on the master channel). Usage
3150 of the HASH() dialplan function, with the key set to the name of the slave
3151 channel, is one approach that will avoid conflicts.
3152 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3154 * func_odbc now allows multiple row results to be retrieved without using
3155 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3156 from the same query by using the name of the function which retrieved the
3157 first row as an argument to ODBC_FETCH().
3158 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3159 dialplan. This function returns the content of the received message.
3160 * Added REPLACE, which searches a given variable name for a set of characters,
3161 then either replaces them with a single character or deletes them.
3162 * Added PASSTHRU, which literally passes the same argument back as its return
3163 value. The intent is to be able to use a literal string argument to
3164 functions that currently require a variable name as an argument.
3165 * HASH-associated variables now can be inherited across channel creation, by
3166 prefixing the name of the hash at assignment with the appropriate number of
3167 underscores, just like variables.
3168 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3169 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3170 whether or not channels that are bridged to the current channel will be
3171 required to have secure signaling and/or media.
3172 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3173 the current channel has secure signaling and/or media.
3174 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3175 "no_media_path" option.
3176 Returns "0" if there is a B channel associated with the call.
3177 Returns "1" if no B channel is associated with the call. The call is either
3178 on hold or is a call waiting call.
3179 * Added option to dialplan function CDR(), the 'f' option
3180 allows for high resolution times for billsec and duration fields.
3181 * FILE() now supports line-mode and writing.
3182 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3183 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3187 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3188 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3189 and is set when a dynamic feature is triggered.
3190 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3191 to dynamically create a new parking lot matching the value this varible is
3193 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3194 features.conf that should be the base for dynamic parkinglots.
3195 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3196 parkinglot should have.
3197 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3198 parkinglot should have.
3199 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3204 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3205 timeout has expired.
3206 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3207 to the caller when an Agent's phone is ringing. This can be used to indicate
3208 to the caller that their call is about to be picked up, which is nice when
3209 one has been on hold for an extened period of time.
3210 * A new config option, penaltymemberslimit, has been added to queues.conf.
3211 When set this option will disregard penalty settings when a queue has too
3213 * A new option, 'I' has been added to both app_queue and app_dial.
3214 By setting this option, Asterisk will not update the caller with
3215 connected line changes or redirecting party changes when they occur.
3216 * A 'relative-periodic-announce' option has been added to queues.conf. When
3217 enabled, this option will cause periodic announce times to be calculated
3218 from the end of announcements rather than from the beginning.
3219 * The autopause option in queues.conf can be passed a new value, "all." The
3220 result is that if a member becomes auto-paused, he will be paused in all
3221 queues for which he is a member, not just the queue that failed to reach
3223 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3224 * The queue logger now allows events to optionally propagate to a file,
3225 even when realtime logging is turned on. Additionally, realtime logging
3226 supports sending the event arguments to 5 individual fields, although it
3227 will fallback to the previous data definition, if the new table layout is
3230 mISDN channel driver (chan_misdn) changes
3231 ----------------------------------------
3232 * Added display_connected parameter to misdn.conf to put a display string
3233 in the CONNECT message containing the connected name and/or number if
3234 the presentation setting permits it.
3235 * Added display_setup parameter to misdn.conf to put a display string
3236 in the SETUP message containing the caller name and/or number if the
3237 presentation setting permits it.
3238 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3239 indicate the dialplan settings are to be obtained from the asterisk
3241 * Made misdn.conf parameter callerid accept the "name" <number> format
3242 used by the rest of the system.
3243 * Made use the nationalprefix and internationalprefix misdn.conf
3244 parameters to prefix any received number from the ISDN link if that
3245 number has the corresponding Type-Of-Number. NOTE: This includes
3246 comparing the incoming call's dialed number against the MSN list.
3247 * Added the following new parameters: unknownprefix, netspecificprefix,
3248 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3249 received number from the ISDN link if that number has the corresponding
3251 * Added new dialplan application misdn_command which permits controlling
3252 the CCBS/CCNR functionality.
3253 * Added new dialplan function mISDN_CC which permits retrieval of various
3254 values from an active call completion record.
3255 * For PTP, you should manually send the COLR of the redirected-to party
3256 for an incomming redirected call if the incoming call could experience
3257 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3258 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3259 if the REDIRECTING(from-num) is not empty.
3260 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3261 option on all of the REDIRECTING statements before dialing the
3262 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3263 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3264 redirecting-to presentation (COLR) when it becomes available.
3265 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3268 thirdparty mISDN enhancements
3269 -----------------------------
3270 mISDN has been modified by Digium, Inc. to greatly expand facility message
3272 * Enhanced COLP support for call diversion and transfer.
3273 * CCBS/CCNR support.
3275 The latest modified mISDN v1.1.x based version is available at:
3276 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3277 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3279 Tagged versions of the modified mISDN code are available under:
3280 http://svn.digium.com/svn/thirdparty/mISDN/tags
3281 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3283 libpri channel driver (chan_dahdi) DAHDI changes
3284 -------------------------------------------
3285 * The channel variable PRIREDIRECTREASON is now just a status variable
3286 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3287 to read and alter the reason.
3288 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3289 redirected-to party for an incomming redirected call if the incoming call
3290 could experience further redirects. Just set the
3291 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3292 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3294 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3295 use the inhibit(i) option on all of the REDIRECTING statements before
3296 dialing the redirected-to party. You still have to set the
3297 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3298 will update the redirecting-to presentation (COLR) when it becomes available.
3299 * Added the ability to ignore calls that are not in a Multiple Subscriber
3300 Number (MSN) list for PTMP CPE interfaces.
3301 * Added dynamic range compression support for dahdi channels. It is
3302 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3303 * Added support for ISDN calling and called subaddress with partial support
3304 for connected line subaddress.
3305 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3306 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3307 to transfer a held call on disconnect similar to an analog phone.
3308 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3309 Will reroute/deflect an outgoing call when receive the message.
3310 Can use the DAHDISendCallreroutingFacility to send the message for the
3312 * Added standard location to add options to chan_dahdi dialing:
3313 Dial(DAHDI/g1[/extension[/options]])
3316 R Reverse charging indication
3317 * Added Reverse Charging Indication (Collect calls) send/receive option.
3318 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3319 Dial(DAHDI/g1/extension/R)
3320 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3321 (requires latest LibPRI)
3322 * Added ability to send/receive keypad digits in the SETUP message.
3323 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3324 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3325 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3326 (requires latest LibPRI)
3327 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3328 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3329 back into the same interface. Tromboned calls happen because of call routing,
3330 call deflection, call forwarding, and call transfer.
3331 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3332 * Added the ability to support call waiting calls. (The SETUP has no B channel
3334 * Added Malicious Call ID (MCID) event to the AMI call event class.
3335 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3337 Asterisk Manager Interface
3338 --------------------------
3339 * The Hangup action now accepts a Cause header which may be used to
3340 set the channel's hangup cause.
3341 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3342 to specify a separate .pem file to hold a private key. By default sslcert
3343 is used to hold both the public and private key.
3344 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3345 for options containing the 'tls' prefix. For example, 'sslenable' is now
3346 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3347 across all .conf files. All affected sample.conf files have been modified to
3348 reflect this change. Previous options such as 'sslenable' still work,
3349 but options with the 'tls' prefix are preferred.
3350 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3351 in a channel. (res_mutestream.so)
3352 * The configuration file manager.conf now supports a channelvars option, which
3353 specifies a list of channel variables to include in each channel-oriented
3355 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3356 and ExtraPriority to allow redirecting the second channel to a different
3357 location than the first.
3358 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3360 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3361 in a MixMonitor recording.
3362 * The 'iax2 show peers' output is now similar to the expected output of
3364 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3366 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3367 AOC-E messages on a channel.
3368 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3369 conform more closely to similar events.
3370 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3372 * Added optional parkinglot variable for park command.
3373 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3374 if CallerIDNum and CallerIDName headers are also present.
3376 Channel Event Logging
3377 ---------------------
3378 * A new interface, CEL, is introduced here. CEL logs single events, much like
3379 the AMI, but it differs from the AMI in that it logs to db backends much
3380 like CDR does; is based on the event subsystem introduced by Russell, and
3381 can share in all its benefits; allows multiple backends to operate like CDR;
3382 is specialized to event data that would be of concern to billing sytems,
3383 like CDR. Backends for logging and accounting calls have been produced,
3384 but a new CDR backend is still in development.
3388 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3389 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3390 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3391 * Multiple files and formats can now be specified in cdr_custom.conf.
3392 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3393 See configs/cdr_syslog.conf.sample for more information.
3394 * A 'sequence' field has been added to CDRs which can be combined with
3395 linkedid or uniqueid to uniquely identify a CDR.
3396 * Handling of billsec and duration field has changed. If your table definition
3397 specifies those fields as float,double or similar they will now be logged with
3398 microsecond accuracy instead of a whole integer.
3400 Calendaring for Asterisk
3401 ------------------------
3402 * A new set of modules were added supporing calendar integration with Asterisk.
3403 Dialplan functions for reading from and writing to calendars are included,
3404 as well as the ability to execute dialplan logic upon calendar event notifications.
3405 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3406 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3407 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
3408 2003 support does not support forms-based authentication).
3410 Call Completion Supplementary Services for Asterisk
3411 ---------------------------------------------------
3412 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
3413 DAHDI/ISDN supports call completion for the following switch types:
3414 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
3415 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
3417 Multicast RTP Support
3418 ---------------------
3419 * A new RTP engine and channel driver have been added which supports Multicast RTP.
3420 The channel driver can be used with the Page application to perform multicast RTP
3421 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
3422 Type can be either basic or linksys.
3423 Destination is the IP address and port for the RTP packets.
3424 Control address is specific to the linksys type and is used for sending the control
3425 packets unique to them.
3427 Security Events Framework
3428 -------------------------
3429 * Asterisk has a new C API for reporting security events. The module res_security_log
3430 sends these events to the "security" logger level. Currently, AMI is the only
3431 Asterisk component that reports security events. However, SIP support will be
3432 coming soon. For more information on the security events framework, see the
3433 "Asterisk Security Framework" section of the Asterisk wiki at
3434 https://wiki.asterisk.org/wiki/x/wgBQ
3435 * SIP support was added in Asterisk 10
3436 * This API now supports IPv6 addresses
3440 * A technology independent fax frontend (res_fax) has been added to Asterisk.
3441 * A spandsp based fax backend (res_fax_spandsp) has been added.
3442 * The app_fax module has been deprecated in favor of the res_fax module and
3443 the new res_fax_spandsp backend.
3444 * The SendFAX and ReceiveFAX applications now send their log messages to a
3445 'fax' logger level, instead of to the generic logger levels. To see these
3446 messages, the system's logger.conf file will need to direct the 'fax' logger
3447 level to one or more destinations; the logger.conf.sample file includes an
3448 example of how to do this. Note that if the 'fax' logger level is *not*
3449 directed to at least one destination, log messages generated by these
3450 applications will be lost, and that if the 'fax' logger level is directed to
3451 the console, the 'core set verbose' and 'core set debug' CLI commands will
3452 have no effect on whether the messages appear on the console or not.
3456 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
3457 Now, in order to enable transmitting silence during record the transmit_silence
3458 option should be used. transmit_silence_during_record remains a valid option, but
3459 defaults to the behavior of the transmit_silence option.
3460 * Addition of the Unit Test Framework API for managing registration and execution
3461 of unit tests with the purpose of verifying the operation of C functions.
3462 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
3463 XMPP text messages to the remote JID.
3464 * Modules.conf has a new option - "require" - that marks a module as critical for
3465 the execution of Asterisk.
3466 If one of the required modules fail to load, Asterisk will exit with a return
3468 * An 'X' option has been added to the asterisk application which enables #exec support.
3469 This allows #exec to be used in asterisk.conf.
3470 * jabber.conf supports a new option auth_policy that toggles auto user registration.
3471 * A new lockconfdir option has been added to asterisk.conf to protect the
3472 configuration directory (/etc/asterisk by default) during reloads.
3473 * The parkeddynamic option has been added to features.conf to enable the creation
3474 of dynamic parkinglots.
3475 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
3476 the reportalarms config option.
3477 * chan_dahdi supports dialing configuring and dialing by device file name.
3478 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3479 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3480 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3481 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3482 Handy for the above name-based syntax as it does not depend on
3483 initialization order.
3484 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3485 significant increase in performance (about 3X) for installations using this switchtype.
3486 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3487 AIS. For more information, please see the Distributed Device State section of the
3488 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3489 * The addition of G.719 pass-through support.
3490 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3491 during device configuration.
3492 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3493 have less than 3 lines on the LCD.
3494 * Realtime now supports database failover. See the sample extconfig.conf for details.
3495 * The addition of improved translation path building for wideband codecs. Sample
3496 rate changes during translation are now avoided unless absolutely necessary.
3497 * The addition of the res_stun_monitor module for monitoring and reacting to network
3498 changes while behind a NAT.
3499 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3500 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3501 These allow support for any Administration. Default is AT&T values.
3505 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3506 optionally accept a filename, to apply the setting only to the code generated from
3507 that source file when Asterisk was built. However, there are some modules in Asterisk
3508 that are composed of multiple source files, so this did not result in the behavior
3509 that users expected. In this version, 'core set debug' and 'core set verbose'
3510 can optionally accept *module* names instead (with or without the .so extension),
3511 which applies the setting to the entire module specified, regardless of which source
3512 files it was built from.
3513 * New 'manager show settings' command showing the current settings loaded from
3515 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3516 the channel hangup request to all channels.
3517 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3519 ------------------------------------------------------------------------------
3520 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3521 ------------------------------------------------------------------------------
projects / asterisk/asterisk.git / blob