1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
13 ------------------------------------------------------------------------------
16 --------------------------
17 * Record application now has an option 'o' which allows 0 to act as an exit
18 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
21 --------------------------
22 * A new function was added: PERIODIC_HOOK. This allows running a periodic
23 dialplan hook on a channel. Any audio generated by this hook will be
24 injected into the call.
27 --------------------------
28 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
29 as the chanprefix parameter if the 'u' option is specified.
32 --------------------------
33 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
34 conference user menus.
36 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
37 menus, bridge settings, and user settings that have been applied by the
38 CONFBRIDGE dialplan function.
40 * The ConfBridge dialplan application now sets a channel variable,
41 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
42 how a channel exited the conference.
44 * Added conference user option 'announce_join_leave_review'. This option
45 implies 'announce_join_leave' with the added effect that the user will
46 be asked if they want to confirm or re-record the recording of their
47 name when entering the conference
50 --------------------------
51 * At exit, the Directory application now sets a channel variable
52 DIRECTORY_RESULT to one of the following based on the reason for exiting:
53 OPERATOR user requested operator by pressing '0' for operator
54 ASSISTANT user requested assistant by pressing '*' for assistant
55 TIMEOUT user pressed nothing and Directory stopped waiting
56 HANGUP user's channel hung up
57 SELECTED user selected a user from the directory and is routed
58 USEREXIT user pressed '#' from the selection prompt to exit
59 FAILED directory failed in a way that wasn't accounted for. Dang.
62 --------------------------
63 * MusicOnHold streams (all modes other than "files") now support wide band
67 --------------------------
68 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
69 and for the channel executing Page respectively.
72 --------------------------
73 * PickupChan now accepts channel uniqueids of channels to pickup.
76 --------------------------
77 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
78 to 'true' (case insensitive), then any Say application (SayNumber,
79 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
80 anticipate DTMF. If DTMF is received, these applications will behave like
81 the background application and jump to the received extension once a match
82 is established or after a short period of inactivity.
85 -------------------------
86 * A new function, MIXMONITOR, has been added to allow access to individual
87 instances of MixMonitor on a channel.
90 -------------------------
91 * Core Show Locks output now includes Thread/LWP ID if the platform
92 supports this feature.
93 * New "logger add channel" and "logger remove channel" CLI commands have
94 been added to allow creation and deletion of dynamic logger channels
95 without configuration changes. These dynamic logger channels will only
96 exist until the next restart of asterisk.
100 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
101 the new AST_SORCERY diaplan function.
105 * The live recording object on recording events now contains a target_uri
106 field which contains the URI of what is being recorded.
108 * The bridge type used when creating a bridge is now a comma separated list of
109 bridge properties. Valid options are: mixing, holding, dtmf_events, and
112 * A channelId can now be provided when creating a channel, either in the
113 uri (POST channels/my-channel-id) or as query parameter. A local channel
114 will suffix the second channel id with ';2' unless provided as query
115 parameter otherChannelId.
117 * A bridgeId can now be provided when creating a bridge, either in the uri
118 (POST bridges/my-bridge-id) or as a query parameter.
120 * A playbackId can be provided when starting a playback, either in the uri
121 (POST channels/my-channel-id/play/my-playback-id) or as a query parameter.
123 * A snoop channel can be started with a snoopId, in the uri or query.
127 * Originate now takes optional parameters ChannelId and OtherChannelId,
128 used to set the UniqueId on creation. The other id is assigned to the
129 second channel when dialing LOCAL, or defaults to appending ;2 if only
130 the single Id is given.
134 * A new set of Alembic scripts has been added for CDR tables. This will create
135 a 'cdr' table with the default schema that Asterisk expects.
139 * A new module, res_hep, has been added, that acts as a generic packet
140 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
141 It can be configured via hep.conf. Other modules can use res_hep to send
142 message traffic to a HEP capture server.
146 * A new module, res_hep_pjsip, has been added that will forward PJSIP
147 message traffic to a HEP capture server. See res_hep for more
152 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
153 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
155 * Added the following new CLI commands:
156 - "pjsip show contacts" - list all current PJSIP contacts.
157 - "pjsip show contact" - show specific information about a current PJSIP
159 - "pjsip show channel" - show detailed information about a PJSIP channel.
163 * A new module, res_pjsip_multihomed handles situations where the system
164 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
165 determines which interface should be used during message sending.
167 res_pjsip_pidf_digium_body_supplement
169 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
170 request body formatting for presence support in Digium phones.
172 res_pjsip_send_to_voicemail
174 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
175 particular headers to transfer a PJSIP channel directly to a particular
176 extension that has VoiceMail. This is intended to be used with Digium
177 phones that support this feature.
179 res_pjsip_outbound_registration
181 * A new CLI command has been added: "pjsip show registrations", which lists
182 all configured PJSIP registrations
185 ------------------------------------------------------------------------------
186 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
187 ------------------------------------------------------------------------------
191 * Added a new module that provides AMI control over MWI within Asterisk,
192 res_mwi_external_ami. Note that this module depends on res_mwi_external;
193 for more information on enabling this module, see res_mwi_external.
194 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
195 the MWIGet/MWIGetComplete events.
197 * The DialStatus field in the DialEnd event can now contain additional
198 statuses that convey how the dial operation terminated. This includes
199 ABORT, CONTINUE, and GOTO.
201 * AMI will now emit security events. A new class authorization has been
202 added in manager.conf for the security events, 'security'. The new events
204 - FailedACL - raised when a request violates an ACL check
205 - InvalidAccountID - raised when a request fails an authentication
206 check due to an invalid account ID
207 - SessionLimit - raised when a request fails due to exceeding the
208 number of allowed concurrent sessions for a service
209 - MemoryLimit - raised when a request fails due to an internal memory
211 - LoadAverageLimit - raised when a request fails because a configured
212 load average limit has been reached
213 - RequestNotAllowed - raised when a request is not allowed by
215 - AuthMethodNotAllowed - raised when a request used an authentication
216 method not allowed by the service
217 - RequestBadFormat - raised when a request is received with bad formatting
218 - SuccessfulAuth - raised when a request successfully authenticates
219 - UnexpectedAddress - raised when a request has a different source address
220 then what is expected for a session already in progress with a service
221 - ChallengeResponseFailed - raised when a request's attempt to authenticate
222 has been challenged, and the request failed the authentication challenge
223 - InvalidPassword - raised when a request provides an invalid password
224 during an authentication attempt
225 - ChallengeSent - raised when an Asterisk service send an authentication
226 challenge to a request
227 - InvalidTransport - raised when a request attempts to use a transport not
228 allowed by the Asterisk service
230 * Bridge related events now have two additional fields: BridgeName and
231 BridgeCreator. BridgeName is a descriptive name for the bridge;
232 BridgeCreator is the name of the entity that created the bridge. This
233 affects the following events: ConfbridgeStart, ConfbridgeEnd,
234 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
235 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
236 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
240 * The Bridge data model now contains the additional fields 'name' and
241 'creator'. The 'name' field conveys a descriptive name for the bridge;
242 the 'creator' field conveys the name of the entity that created the bridge.
243 This affects all responses to HTTP requests that return a Bridge data model
244 as well as all event derived data models that contain a Bridge data model.
245 The POST /bridges operation may now optionally specify a name to give to
246 the bridge being created.
248 * Added a new ARI resource 'mailboxes' which allows the creation and
249 modification of mailboxes managed by external MWI. Modules res_mwi_external
250 and res_stasis_mailbox must be enabled to use this resource. For more
251 information on external MWI control, see res_mwi_external.
253 * Added new events for externally initiated transfers. The event
254 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
255 of a bridge in the ARI controlled application to the dialplan; the
256 BridgeAttendedTransfer event is raised when a channel initiates an
257 attended transfer of a bridge in the ARI controlled application to the
260 * Channel variables may now be specified as a body parameter to the
261 POST /channels operation. The 'variables' key in the JSON is interpreted
262 as a sequence of key/value pairs that will be added to the created channel
263 as channel variables. Other parameters in the JSON body are treated as
264 query parameters of the same name.
268 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
269 automatically handled by the HTTP server if a request is received with a
270 Transfer-Encoding type of "chunked".
274 * Path support has been added with the 'support_path' option in registration
277 * A 'debug' option has been added to the globals section that will allow
278 sip messages to be logged.
280 * A 'set_var' option has been added to endpoints that will automatically
281 set the desired variable(s) on a channel created for that endpoint.
283 * Several new tables and columns have been added to the realtime schema for
284 the res_pjsip related modules. See the UPGRADE.txt notes for updating
289 * A new module, res_mwi_external, has been added to Asterisk. This module
290 acts as a base framework that other modules can build on top of to allow
291 an external system to control MWI within Asterisk. For implementations
292 that make use of res_mwi_external, see res_mwi_external_ami and
293 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
294 that may produce MWI themselves, such as app_voicemail. res_mwi_external
295 and other modules that depend on it cannot be built or loaded with
296 app_voicemail present.
300 * DNS functionality will now automatically be enabled if the system configured
301 nameservers can be retrieved. If the system configured nameservers can not be
302 retrieved the functionality will resort to using system resolution. Functionalty
303 such as SRV records and failover will not be available if system resolution
306 ------------------------------------------------------------------------------
307 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
308 ------------------------------------------------------------------------------
313 Asterisk 12 is a standard release of the Asterisk project. As such, the
314 focus of development for this release was on core architectural changes and
315 major new features. This includes:
316 * A more flexible bridging core based on the Bridging API
317 * A new internal message bus, Stasis
318 * Major standardization and consistency improvements to AMI
319 * Addition of the Asterisk RESTful Interface (ARI)
320 * A new SIP channel driver, chan_pjsip
321 In addition, as the vast majority of bridging in Asterisk was migrated to the
322 Bridging API used by ConfBridge, major changes were made to most of the
323 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
325 Specifications have been written for the affected interfaces. These
326 specifications are available on the Asterisk wiki:
327 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
328 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
329 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
331 It is *highly* recommended that anyone migrating to Asterisk 12 read the
332 information regarding its release both in this file and in the accompanying
333 UPGRADE.txt file. More detailed information on the major changes can be found
334 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
339 * Added build option DISABLE_INLINE. This option can be used to work around a
340 bug in gcc. For more information, see
341 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
343 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
344 the CHANNEL_TRACE build option were incompatible with the new bridging
347 * Asterisk now optionally uses libxslt to improve XML documentation generation
348 and maintainability. If libxslt is not available on the system, some XML
349 documentation will be incomplete.
351 * Asterisk now depends on libjansson. If a package of libjansson is not
352 available on your distro, please see http://www.digip.org/jansson/.
354 * Asterisk now depends on libuuid and, optionally, uriparser. It is
355 recommended that you install uriparser, even if it is optional.
357 * The new SIP stack and channel driver uses a particular version of PJSIP.
358 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
359 configuring and installing PJSIP for usage with Asterisk.
361 * Optional API was re-implemented to be more portable, and no longer requires
362 weak reference support from the compiler. The build option OPTIONAL_API may
363 be disabled to disable Optional API support.
370 * Along with AgentRequest, this application has been modified to be a
371 replacement for chan_agent. The act of a channel calling the AgentLogin
372 application places the channel into a pool of agents that can be
373 requested by the AgentRequest application. Note that this application, as
374 well as all other agent related functionality, is now provided by the
375 app_agent_pool module. See chan_agent and AgentRequest for more information.
377 * This application no longer performs agent authentication. If authentication
378 is desired, the dialplan needs to perform this function using the
379 Authenticate or VMAuthenticate application or through an AGI script before
382 * If this application is called and the agent is already logged in, the
383 dialplan will continue exection with the AGENT_STATUS channel variable set
384 to ALREADY_LOGGED_IN.
386 * The agents.conf schema has changed. Rather than specifying agents on a
387 single line in comma delineated fashion, each agent is defined in a separate
388 context. This allows agents to use the power of context templates in their
391 * A number of parameters from agents.conf have been removed. This includes
392 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
393 urlprefix, and savecallsin. These options were obsoleted by the move from
394 a channel driver model to the bridging/application model provided by
399 * A new application, this will request a logged in agent from the pool and
400 bridge the requested channel with the channel calling this application.
401 Logged in agents are those channels that called the AgentLogin application.
402 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
403 application will be set with an appropriate error value.
407 * This application has been removed. It was a holdover from when
408 AgentCallbackLogin was removed.
412 * Added support for additional Ademco DTMF signalling formats, including
413 Express 4+1, Express 4+2, High Speed and Super Fast.
415 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
416 call time, in milliseconds, to run the application.
418 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
419 maximum number of times to retry the call.
421 * Added a new configuration option answait. If set, the AlarmReceiver
422 application will wait the number of milliseconds specified by answait
423 after the channel has answered. Valid values range between 500
424 milliseconds and 10000 milliseconds.
426 * Added configuration option no_group_meta. If enabled, grouping of metadata
427 information in the AlarmReceiver log file will be skipped.
431 * It is now no longer possible to bypass updating the CDR on the channel
432 when answering. CDRs reflect the state of the channel and will always
433 reflect the time they were Answered.
437 * A new application in Asterisk, this will place the calling channel
438 into a holding bridge, optionally entertaining them with some form of
439 media. Channels participating in a holding bridge do not interact with
440 other channels in the same holding bridge. Optionally, however, a channel
441 may join as an announcer. Any media passed from an announcer channel is
442 played to all channels in the holding bridge. Channels leave a holding
443 bridge either when an optional timer expires, or via the ChannelRedirect
444 application or AMI Redirect action.
448 * All participants in a bridge can now be kicked out of a conference room
449 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
450 command, i.e., 'confbridge kick <conference> all'
452 * CLI output for the 'confbridge list' command has been improved. When
453 displaying information about a particular bridge, flags will now be shown
454 for the participating users indicating properties of that user.
456 * The ConfbridgeList event now contains the following fields: WaitMarked,
457 EndMarked, and Waiting. This displays additional properties about the
458 user's profile, as well as whether or not the user is waiting for a
459 Marked user to enter the conference.
461 * Added a new option for conference recording, record_file_append. If enabled,
462 when the recording is stopped and then re-started, the existing recording
463 will be used and appended to.
465 * ConfBridge now has the ability to set the language of announcements to the
466 conference. The language can be set on a bridge profile in confbridge.conf
467 or by the dialplan function CONFBRIDGE(bridge,language)=en.
471 * The channel variable CPLAYBACKSTATUS may now return the value
472 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
473 such as AMI. See the AMI action ControlPlayback for more information.
477 * Added the 'a' option, which allows the caller to enter in an additional
478 alias for the user in the directory. This option must be used in conjunction
479 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
480 specified in voicemail.conf.
484 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
485 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
486 containing the unique ID of the bridge that the channel happens to be in.
490 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
491 for more information.
493 * Variables are no longer purged from the original CDR. See the 'v' option for
496 * The 'A' option has been removed. The Answer time on a CDR is never updated
499 * The 'd' option has been removed. The disposition on a CDR is a function of
500 the state of the channel and cannot be altered.
502 * The 'D' option has been removed. Who the Party B is on a CDR is a function
503 of the state of the respective channels involved in the CDR and cannot be
506 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
507 such that the start time and, if applicable, the answer time was updated.
508 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
509 'r' option now triggers the Reset, setting the start time (and answer time
510 if applicable) to the current time. Note that the 'a' option still sets
511 the answer time to the current time if the channel was already answered.
513 * The 's' option has been removed. A variable can be set on the original CDR
514 if desired using the CDR function, and removed from a forked CDR using the
517 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
518 longer applies in the CDR engine.
520 * The 'v' option now prevents the copy of the variables from the original CDR
521 to the forked CDR. Previously the variables were always copied but were
522 removed from the original. This was changed as removing variables from a CDR
523 can have unintended side effects - this option allows the user to prevent
524 propagation of variables from the original to the forked without modifying
529 * Added the 'n' option to MeetMe to prevent application of the DENOISE
530 function to a channel joining a conference. Some channel drivers that vary
531 the number of audio samples in a voice frame will experience significant
532 quality problems if a denoiser is attached to the channel; this option gives
533 them the ability to remove the denoiser without having to unload func_speex.
537 * The 'b' option now includes conferences as well as sounds played to the
540 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
541 running during a transfer. If a MixMonitor is started on a channel,
542 the MixMonitor will continue to record the audio passing through the
543 channel even in the presence of transfers.
547 * The NoCDR application is deprecated. Please use the CDR_PROP function to
550 * While the NoCDR application will prevent CDRs for a channel from being
551 propagated to registered CDR backends, it will not prevent that data from
552 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
553 function that enables CDRs on a channel will restore those records that have
554 not yet been finalized.
558 * The app_parkandannounce module has been removed. The application
559 ParkAndAnnounce is now provided by the res_parking module. See the
560 res_parking changes for more information.
564 * Added queue available hint. The hint can be added to the dialplan using the
565 following syntax: exten,hint,Queue:{queue_name}_avail
566 For example, if the name of the queue is 'markq':
567 exten => 8501,hint,Queue:markq_avail
568 This will report 'InUse' if there are no logged in agents or no free agents.
569 It will report 'Idle' when an agent is free.
571 * Queues now support a hint for member paused state. The hint uses the form
572 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
573 are the name of the queue and the name of the member to subscribe to,
574 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
575 Members will show as In Use when paused.
577 * The configuration options eventwhencalled and eventmemberstatus have been
578 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
579 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
580 sent. The "Variable" fields will also no longer exist on the Agent* events.
581 These events can be filtered out from a connected AMI client using the
582 eventfilter setting in manager.conf.
584 * The queue log now differentiates between blind and attended transfers. A
585 blind transfer will result in a BLINDTRANSFER message with the destination
586 context and extension. An attended transfer will result in an
587 ATTENDEDTRANSFER message. This message will indicate the method by which
588 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
589 for running an application on a bridge or channel, or "LINK" for linking
590 two bridges together with local channels. The queue log will also now detect
591 externally initiated blind and attended transfers and record the transfer
594 * When performing queue pause/unpause on an interface without specifying an
595 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
596 least one member of any queue exists for that interface.
598 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
599 for realtime queue log entries.
603 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
604 CDRs when they were previously disabled on a channel.
606 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
607 backends occurs on an as-needed basis in order to preserve linkedid
608 propagation and other needed behavior.
612 * A new application, this is similar to SayAlpha except that it supports
613 case sensitive playback of the specified characters. For example,
614 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
618 * This application is deprecated in favor of CHANNEL(amaflags).
622 * The SendDTMF application will now accept 'W' as valid input. This will cause
623 the application to delay one second while streaming DTMF.
627 * A new application in Asterisk 12, this hands control of the channel calling
628 the application over to an external system. Currently, external systems
629 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
633 * UserEvent will now handle duplicate keys by overwriting the previous value
636 * In addition to AMI, UserEvent invocations will now be distributed to any
637 interested Stasis applications.
641 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
642 system as mailbox@context. The rest of the system cannot add @default
643 to mailbox identifiers for app_voicemail that do not specify a context
644 any longer. It is a mailbox identifier format that should only be
645 interpreted by app_voicemail.
647 * The voicemail.conf configuration file now has an 'alias' configuration
648 parameter for use with the Directory application. The voicemail realtime
649 database table schema has also been updated with an 'alias' column.
654 * Pass through support has been added for both VP8 and Opus.
656 * Added format attribute negotiation for the Opus codec. Format attribute
657 negotiation is provided by the res_format_attr_opus module.
662 * Masquerades as an operation inside Asterisk have been effectively hidden
663 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
664 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
665 dropping of frame/audio hooks, and other internal implementation details
666 that users had to deal with. This fundamental change has large implications
667 throughout the changes documented for this version. For more information
668 about the new core architecture of Asterisk, please see the Asterisk wiki.
670 * Multiple parties in a bridge may now be transferred. If a participant in a
671 multi-party bridge initiates a blind transfer, a Local channel will be used
672 to execute the dialplan location that the transferer sent the parties to. If
673 a participant in a multi-party bridge initiates an attended transfer,
674 several options are possible. If the attended transfer results in a transfer
675 to an application, a Local channel is used. If the attended transfer results
676 in a transfer to another channel, the resulting channels will be merged into
679 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
680 driver specific. If the channel variable is set on the transferrer channel,
681 the sound will be played to the target of an attended transfer.
683 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
684 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
685 listed. Any more peers in the bridge will not be included in the list.
686 BRIDGEPEER is not valid in holding bridges like parking since those channels
687 do not talk to each other even though they are in a bridge.
689 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
690 and will contain a value if the BRIDGEPEER's channel driver supports it.
692 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
693 was responsible for an attended transfer in a similar fashion to
696 * Modules using the Configuration Framework or Sorcery must have XML
697 configuration documentation. This configuration documentation is included
698 with the rest of Asterisk's XML documentation, and is accessible via CLI
699 commands. See the CLI changes for more information.
701 AMI (Asterisk Manager Interface)
703 * Major changes were made to both the syntax as well as the semantics of the
704 AMI protocol. In particular, AMI events have been substantially improved
705 in this version of Asterisk. For more information, please see the AMI
706 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
708 * AMI events that reference a particular channel or bridge will now always
709 contain a standard set of fields. When multiple channels or bridges are
710 referenced in an event, fields for at least some subset of the channels
711 and bridges in the event will be prefixed with a descriptive name to avoid
712 name collisions. See the AMI event documentation on the Asterisk wiki for
715 * The CLI command 'manager show commands' no longer truncates command names
716 longer than 15 characters and no longer shows authorization requirement
717 for commands. 'manager show command' now displays the privileges needed
718 for using a given manager command instead.
720 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
721 peer in its response if the peer has a subscribe context set.
723 * The SIPqualifypeer action now acknowledges the request once it has
724 established that the request is against a known peer. It also issues a new
725 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
727 * The PlayDTMF action now supports an optional 'Duration' parameter. This
728 specifies the duration of the digit to be played, in milliseconds.
730 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
731 updates when changes occur instead of requiring the use of pollmailboxes.
733 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
734 AMI client to manipulate audio currently being played back on a channel. The
735 supported operations depend on the application being used to send audio to
736 the channel. When the audio playback was initiated using the ControlPlayback
737 application or CONTROL STREAM FILE AGI command, the audio can be paused,
738 stopped, restarted, reversed, or skipped forward. When initiated by other
739 mechanisms (such as the Playback application), the audio can be stopped,
740 reversed, or skipped forward.
742 * Channel related events now contain a snapshot of channel state, adding new
743 fields to many of these events.
745 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
746 in a future release. Please use the common 'Exten' field instead.
748 * The AMI event 'UserEvent' from app_userevent now contains the channel state
749 fields. The channel state fields will come before the body fields.
751 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
752 'UnParkedCall' have changed significantly in the new res_parking module.
754 The 'Channel' and 'From' headers are gone. For the channel that was parked
755 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
756 has a number of fields associated with it. The old 'Channel' header relayed
757 the same data as the new 'ParkeeChannel' header.
759 The 'From' field was ambiguous and changed meaning depending on the event.
760 for most of these, it was the name of the channel that parked the call
761 (the 'Parker'). There is no longer a header that provides this channel name,
762 however the 'ParkerDialString' will contain a dialstring to redial the
763 device that parked the call.
765 On UnParkedCall events, the 'From' header would instead represent the
766 channel responsible for retrieving the parkee. It receives a channel
767 snapshot labeled 'Retriever'. The 'from' field is is replaced with
770 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
772 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
773 fashion has changed the field names 'StartExten' and 'StopExten' to
774 'StartSpace' and 'StopSpace' respectively.
776 * The deprecated use of | (pipe) as a separator in the channelvars setting in
777 manager.conf has been removed.
779 * Channel Variables conveyed with a channel no longer contain the name of the
780 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
781 ChanVariable: bar=baz. When multiple channels are present in a single AMI
782 event, the various ChanVariable fields will contain a suffix that specifies
783 which channel they correspond to.
785 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
786 event always conveys the AMI event for a particular channel.
788 * All 'Reload' events have been consolidated into a single event type. This
789 event will always contain a Module field specifying the name of the module
790 and a Status field denoting the result of the reload. All modules now issue
791 this event when being reloaded.
793 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
794 fail to receive this event due to being connected after modules have loaded.
795 AMI connections that want to know when Asterisk is ready should listen for
796 the 'FullyBooted' event.
798 * app_fax now sends the same send fax/receive fax events as res_fax. The
799 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
800 now the 'ReceiveFAX' event.
802 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
803 'MusicOnHoldStop'. The sub type field has been removed.
805 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
806 carrier for another protocol.
808 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
809 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
810 to the specific channel. 'Both' may be specified to play a tone to both
811 channels. The old 'yes' option is still accepted as a way of playing the
812 tone to Channel2 only.
814 * The AMI 'Status' response event to the AMI Status action replaces the
815 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
816 indicate what bridge the channel is currently in.
818 * The AMI 'Hold' event has been moved out of individual channel drivers, into
819 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
822 * The AMI events in app_queue have been made more consistent with each other.
823 Events that reference channels (QueueCaller* and Agent*) will show
824 information about each channel. The (infamous) 'Join' and 'Leave' AMI
825 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
827 * The 'MCID' AMI event now publishes a channel snapshot when available and
828 its non-channel-snapshot parameters now use either the "MCallerID" or
829 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
830 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
831 parameters in the channel snapshot.
833 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
834 'AgentLogin' and 'AgentLogoff' respectively.
836 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
837 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
839 * 'ChannelUpdate' events have been removed.
841 * All AMI events now contain a 'SystemName' field, if available.
843 * Local channel optimization is now conveyed in two events:
844 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
845 when the Local channel driver begins attempting to optimize itself out of
846 the media path; the End event is sent after the channel halves have
847 successfully optimized themselves out of the media path.
849 * Local channel information in events is now prefixed with 'LocalOne' and
850 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
851 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
852 and 'LocalOptimizationEnd' events.
854 * The option 'allowmultiplelogin' can now be set or overriden in a particular
855 account. When set in the general context, it will act as the default
856 setting for defined accounts.
858 * The 'BridgeAction' event was removed. It technically added no value, as the
859 Bridge Action already receives confirmation of the bridge through a
860 successful completion Event.
862 * The 'BridgeExec' events were removed. These events duplicated the events that
863 occur in the Briding API, and are conveyed now through BridgeCreate,
864 BridgeEnter, and BridgeLeave events.
866 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
867 previous versions. They now report all SR/RR packets sent/received, and
868 have been restructured to better reflect the data sent in a SR/RR. In
869 particular, the event structure now supports multiple report blocks.
871 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
872 raised when a blind transfer/attended transfer completes successfully.
873 They contain information about the transfer that just completed, including
874 the location of the transfered channel.
876 * Added a 'security' class to AMI which outputs the required fields for
877 security messages similar to the log messages from res_security_log
879 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
880 that describes the status value in a human readable string.
882 CDR (Call Detail Records)
884 * Significant changes have been made to the behavior of CDRs. The CDR engine
885 was effectively rewritten and built on the Stasis message bus. For a full
886 definition of CDR behavior in Asterisk 12, please read the specification
887 on the Asterisk wiki (wiki.asterisk.org).
889 * CDRs will now be created between all participants in a bridge. For each
890 pair of channels in a bridge, a CDR is created to represent the path of
891 communication between those two endpoints. This lets an end user choose who
892 to bill for what during bridge operations with multiple parties.
894 * The duration, billsec, start, answer, and end times now reflect the times
895 associated with the current CDR for the channel, as opposed to a cumulative
896 measurement of all CDRs for that channel.
898 * When a CDR is dispatched, user defined CDR variables from both parties are
899 included in the resulting CDR. If both parties have the same variable, only
900 the Party A value is provided.
902 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
903 information regarding the CDR engine is logged as verbose messages. This
904 option should only be used if the behavior of the CDR engine needs to be
907 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
908 normally configured in cdr.conf.
910 * Added CLI command 'cdr show active {channel}'. When {channel} is not
911 specified, this command provides a summary of the channels with CDR
912 information and their statistics. When {channel} is specified, it shows
913 detailed information about all records associated with {channel}.
915 CEL (Channel Event Logging)
917 * CEL has undergone significant rework in Asterisk 12, and is now built on the
918 Stasis message bus. Please see the specification for CEL on the Asterisk
919 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
922 * The 'extra' field of all CEL events that use it now consists of a JSON blob
923 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
925 * BLINDTRANSFER events now report the transferee bridge unique
926 identifier, extension, and context in a JSON blob as the extra string
927 instead of the transferee channel name as the peer.
929 * ATTENDEDTRANSFER events now report the peer as NULL and additional
930 information in the 'extra' string as a JSON blob. For transfers that occur
931 between two bridged channels, the 'extra' JSON blob contains the primary
932 bridge unique identifier, the secondary channel name, and the secondary
933 bridge unique identifier. For transfers that occur between a bridged channel
934 and a channel running an app, the 'extra' JSON blob contains the primary
935 bridge unique identifier, the secondary channel name, and the app name.
937 * LOCAL_OPTIMIZE events have been added to convey local channel
938 optimizations with the record occurring for the semi-one channel and
939 the semi-two channel name in the peer field.
941 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
942 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
943 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
944 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
945 regardless of whether or not that bridge happens to contain multiple
950 * When compiled with '--enable-dev-mode', the astobj2 library will now add
951 several CLI commands that allow for inspection of ao2 containers that
952 register themselves with astobj2. The CLI commands are 'astobj2 container
953 dump', 'astobj2 container stats', and 'astobj2 container check'.
955 * Added specific CLI commands for bridge inspection. This includes 'bridge
956 show all', which lists all bridges in the system, and 'bridge show {id}',
957 which provides specific information about a bridge.
959 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
960 ejecting the channels currently in the bridge. If the channels cannot
961 continue in the dialplan or application that put them in the bridge, they
964 * Added command 'bridge kick'. This will eject a single channel from a bridge.
966 * Added commands to inspect and manipulate the registered bridge technologies.
967 This include 'bridge technology show', which lists the registered bridge
968 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
969 which controls whether or not a registered bridge technology can be used
970 during smart bridge operations. If a technology is suspended, it will not
971 be used when a bridge technology is picked for channels; when unsuspended,
972 it can be used again.
974 * The command 'config show help {module} {type} {option}' will show
975 configuration documentation for modules with XML configuration
976 documentation. When {module}, {type}, and {option} are omitted, a listing
977 of all modules with registered documentation is displayed. When {module}
978 is specified, a listing of all configuration types for that module is
979 displayed, along with their synopsis. When {module} and {type} are
980 specified, a listing of all configuration options for that type are
981 displayed along with their synopsis. When {module}, {type}, and {option}
982 are specified, detailed information for that configuration option is
985 * Added 'core show sounds' and 'core show sound' CLI commands. These display
986 a listing of all installed media sounds available on the system and
987 detailed information about a sound, respectively.
989 * 'xmldoc dump' has been added. This CLI command will dump the XML
990 documentation DOM as a string to the specified file. The Asterisk core
991 will populate certain XML elements pulled from the source files with
992 additional run-time information; this command lets a user produce the
993 XML documentation with all information.
997 * Parking has been pulled from core and placed into a separate module called
998 res_parking. See Parking changes below for more details. Configuration for
999 parking should now be performed in res_parking.conf. Configuration for
1000 parking in features.conf is now unsupported.
1002 * Core attended transfers now have several new options. While performing an
1003 attended transfer, the transferer now has the following options:
1004 - *1 - cancel the attended transfer (configurable via atxferabort)
1005 - *2 - complete the attended transfer, dropping out of the call
1006 (configurable via atxfercomplete)
1007 - *3 - complete the attended transfer, but stay in the call. This will turn
1008 the call into a multi-party bridge (configurable via atxferthreeway)
1009 - *4 - swap to the other party. Once an attended transfer has begun, this
1010 options may be used multiple times (configurable via atxferswap)
1012 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1013 must be on the channel initiating the transfer to have any effect.
1015 * The BRIDGE_FEATURES channel variable would previously only set features for
1016 the calling party and would set this feature regardless of whether the
1017 feature was in caps or in lowercase. Use of a caps feature for a letter
1018 will now apply the feature to the calling party while use of a lowercase
1019 letter will apply that feature to the called party.
1021 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1023 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1024 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1025 activated the dynamic feature.
1027 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1028 only on the channel executing the dynamic feature. Executing a dynamic
1029 feature on the bridge peer in a multi-party bridge will execute it on all
1030 peers of the activating channel.
1032 * You can now have the settings for a channel updated using the FEATURE()
1033 and FEATUREMAP() functions inherited to child channels by setting
1034 FEATURE(inherit)=yes.
1036 * automixmon now supports additional channel variables from automon including:
1037 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1038 and TOUCH_MIXMONITOR_MESSAGE_STOP
1040 * A new general features.conf option 'recordingfailsound' has been added which
1041 allowssetting a failure sound for a user tries to invoke a recording feature
1042 such as automon or automixmon and it fails.
1044 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1045 features.c for atxferdropcall=no to work properly. This option now just
1050 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1051 be used. Given that this can cause the Asterisk log files to grow quickly,
1052 this option should only be used if an external mechanism for log management
1057 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1058 will store the path information for that peer when it registers. Realtime
1059 tables can also use the 'supportpath' field to enable Path header support.
1061 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1062 objectIdentifier. This maps to the supportpath option in sip.conf.
1066 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1067 provides modules a useful abstraction on top of the many storage mechanisms
1068 in Asterisk, including the Asterisk Database, static configuration files,
1069 static Realtime, and dynamic Realtime. It also provides a caching service.
1070 Users can configure a hierarchy of data storage layers for specific modules
1073 * All future modules which utilize Sorcery for object persistence must have a
1074 column named "id" within their schema when using the Sorcery realtime module.
1075 This column must be able to contain a string of up to 128 characters in length.
1077 Security Events Framework
1079 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1080 the "seconds-microseconds" format that it was using previously.
1084 * The Stasis message bus is a publish/subscribe message bus internal to
1085 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1086 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1087 Stasis can be configured in stasis.conf. Note that these parameters operate
1088 at a very low level in Asterisk, and generally will not require changes.
1092 * When a channel driver is configured to enable jiterbuffers, they are now
1093 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1094 is already set for that channel when it enters, such as by the JITTERBUFFER
1095 function, then the existing jitterbuffer will be used and the one set by
1096 the channel driver will not be applied.
1100 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1101 dialplan applications provided by the app_agent_pool module. Agents are
1102 connected with callers using the new AgentRequest dialplan application.
1103 The Agents:<agent-id> device state is available to monitor the status of an
1104 agent. See agents.conf.sample for valid configuration options.
1106 * The updatecdr option has been removed. Altering the names of channels on a
1107 CDR is not supported - the name of the channel is the name of the channel,
1108 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1109 has also been removed, for the same reason.
1111 * The endcall and enddtmf configuration options are removed. Use the
1112 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1113 channel before calling AgentLogin.
1117 * chan_bridge has been removed. Its functionality has been incorporated
1118 directly into the ConfBridge application itself.
1122 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1123 of the specified span and its B-channels. Note that this command should
1124 only be used if you understand the risks it entails.
1126 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1127 A range of channels can be specified to be destroyed. Note that this command
1128 should only be used if you understand the risks it entails.
1130 * Added the CLI command 'dahdi create channels'. A range of channels can be
1131 specified to be created, or the keyword 'new' can be used to add channels
1134 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1135 the exact configured mailbox name. For app_voicemail mailboxes this is
1138 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1142 * IPv6 support has been added. We are now able to bind to and
1143 communicate using IPv6 addresses.
1147 * The /b option has been removed.
1149 * chan_local moved into the system core and is no longer a loadable module.
1153 * Added general support for busy detection.
1155 * Added ECAM command support for Sony Ericsson phones.
1159 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1160 SIP stack. A collection of resource modules provides the bulk of the SIP
1161 functionality. For more information on the new SIP channel driver, see
1162 https://wiki.asterisk.org/wiki/x/JYGLAQ
1166 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1167 using the 'supportpath' setting, either on a global basis or on a peer basis.
1168 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1169 set of proxies by using a pre-loaded route-set defined by the Path headers in
1170 the REGISTER request. See Realtime updates for more configuration information.
1172 * The SIP_CODEC family of variables may now specify more than one codec. Each
1173 codec must be separated by a comma. The first codec specified is the
1174 preferred codec for the offer. This allows a dialplan writer to specify both
1175 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1177 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1178 in the core, and can be filtered out using the 'eventfilter' parameter
1181 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1182 codecs configured for a peer instead of the requested codec.
1184 * The option "register_retry_403" has been added to chan_sip to work around
1185 servers that are known to erroneously send 403 in response to valid
1186 REGISTER requests and allows Asterisk to continue attepmting to connect.
1190 * Added the 'immeddialkey' parameter. If set, when the user presses the
1191 configured key the already entered number will be immediately dialed. This
1192 is useful when the dialplan allows for variable length pattern matching.
1193 Valid options are '*' and '#'.
1195 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1196 milliseconds) before a call forward is considered to not be answered.
1198 * The 'serviceurl' parameter allows Service URLs to be attached to line
1207 * The password option has been disabled, as the AgentLogin application no
1208 longer provides authentication.
1212 * Due to changes in the Asterisk core, this function is no longer needed to
1213 preserve a MixMonitor on a channel during transfer operations and dialplan
1214 execution. It is effectively obsolete.
1218 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1219 deprecated. Use the CHANNEL function instead to access these attributes.
1221 * The 'l' option has been removed. When reading a CDR attribute, the most
1222 recent record is always used. When writing a CDR attribute, all non-finalized
1225 * The 'r' option has been removed, for the same reason as the 'l' option.
1227 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1232 * A new function CDR_PROP has been added. This function lets you set properties
1233 on a channel's active CDRs. This function is write-only. Properties accept
1234 boolean values to set/clear them on the channel's CDRs. Valid properties
1236 - 'party_a' - make this channel the preferred Party A in any CDR between two
1237 channels. If two channels have this property set, the creation time of the
1238 channel is used to determine who is Party A. Note that dialed channels are
1239 never Party A in a CDR.
1240 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1241 application when set to True, and analogous to the 'e' option in ResetCDR
1246 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1247 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1248 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1251 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1252 string, i.e., [[context],extension],priority. If set on a channel, if a
1253 channel leaves a bridge but is not hung up it will resume dialplan execution
1258 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1259 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1260 The value of this setting is ignored when disabled is used for the argument.
1264 * A new function provided by chan_pjsip, this function can be used in
1265 conjunction with the Dial application to construct a dial string that will
1266 dial all contacts on an Address of Record associated with a chan_pjsip
1271 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1272 outbound channel prior to dialing.
1276 * Redirecting reasons can now be set to arbitrary strings. This means
1277 that the REDIRECTING dialplan function can be used to set the redirecting
1278 reason to any string. It also allows for custom strings to be read as the
1279 redirecting reason from SIP Diversion headers.
1283 * The SPEECH_ENGINE function now supports read operations. When read from, it
1284 will return the current value of the requested attribute.
1288 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1289 system as mailbox@context. The rest of the system cannot add @default
1290 to mailbox identifiers for app_voicemail that do not specify a context
1291 any longer. It is a mailbox identifier format that should only be
1292 interpreted by app_voicemail.
1298 res_agi (Asterisk Gateway Interface)
1300 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1302 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1305 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1306 will start the playback of the audio at the position specified. It will
1307 also return the final position of the file in 'endpos'.
1309 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1310 channel variable if the user stopped the file playback or if a remote
1311 entity stopped the playback. If neither stopped the playback, it will
1312 indicate the overall success/failure of the playback. If stopped early,
1313 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1316 * The SAY ALPHA command now accepts an additional parameter to control
1317 whether it specifies the case of uppercase, lowercase, or all letters to
1318 provide functionality similar to SayAlphaCase.
1320 res_ari (Asterisk RESTful Interface) (and others)
1322 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1323 control telephony primitives in Asterisk by remote client. This includes
1324 channels, bridges, endpoints, media, and other fundamental concepts. Users
1325 of ARI can develop their own communications applications, controlling
1326 multiple channels using an HTTP RESTful interface and receiving JSON events
1327 about the objects via a WebSocket connection. ARI can be configured in
1328 Asterisk via ari.conf. For more information on ARI, see
1329 https://wiki.asterisk.org/wiki/x/0YCLAQ
1333 * Parking has been extracted from the Asterisk core as a loadable module,
1334 res_parking. Configuration for parking is now provided by res_parking.conf.
1335 Configuration through features.conf is no longer supported.
1337 * res_parking uses the configuration framework. If an invalid configuration is
1338 supplied, res_parking will fail to load or fail to reload. Previously,
1339 invalid configurations would generally be accepted, with certain errors
1340 resulting in individually disabled parking lots.
1342 * Parked calls are now placed in bridges. While this is largely an
1343 architectural change, it does have implications on how channels in a parking
1344 lot are viewed. For example, commands that display channels in bridges will
1345 now also display the channels in a parking lot.
1347 * The order of arguments for the new parking applications have been modified.
1348 Timeout and return context/exten/priority are now implemented as options,
1349 while the name of the parking lot is now the first parameter. See the
1350 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1351 in-depth information as well as syntax.
1353 * Extensions are by default no longer automatically created in the dialplan to
1354 park calls or pickup parked calls. Generation of dialplan extensions can be
1355 enabled using the 'parkext' configuration option.
1357 * ADSI functionality for parking is no longer supported. The 'adsipark'
1358 configuration option has been removed as a result.
1360 * The PARKINGSLOT channel variable has been deprecated in favor of
1361 PARKING_SPACE to match the naming scheme of the new system.
1363 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1364 channel even when the configuration option 'comebactoorigin' is enabled.
1366 * A new CLI command 'parking show' has been added. This allows a user to
1367 inspect the parking lots that are currently in use.
1368 'parking show <parkinglot>' will also show the parked calls in a specific
1371 * The CLI command 'parkedcalls' is now deprecated in favor of
1372 'parking show <parkinglot>'.
1374 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1375 can be used to get a list of parked calls for a specific parking lot.
1377 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1378 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1379 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1380 longer a required argument.
1382 * The ParkAndAnnounce application is now provided through res_parking instead
1383 of through the separate app_parkandannounce module.
1385 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1386 by default. Instead, it will follow the timeout rules of the parking lot. The
1387 old behavior can be reproduced by using the 'c' option.
1389 * Dynamic parking lots will now fail to be created under the following
1391 - if the parking lot specified by PARKINGDYNAMIC does not exist
1392 - if they require exclusive park and parkedcall extensions which overlap
1393 with existing parking lots.
1395 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1396 currently contain no calls. Dynamic parking lots containing parked calls
1397 will persist through the reloads without alteration.
1399 * If 'parkext_exclusive' is set for a parking lot and that extension is
1400 already in use when that parking lot tries to register it, this is now
1401 considered a parking system configuration error. Configurations which do
1402 this will be rejected.
1404 * Added channel variable PARKER_FLAT. This contains the name of the extension
1405 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1406 comebacktoorigin is disabled, but the dialplan or an external control
1407 mechanism wants to use the extension in the park-dial context that was
1408 generated to re-dial the parker on timeout.
1410 res_pjsip (and many others)
1412 * A large number of resource modules make up the SIP stack based on pjsip.
1413 The chan_pjsip channel driver users these resource modules to provide
1414 various SIP functionality in Asterisk. The majority of configuration for
1415 these modules is performed in pjsip.conf. Other modules may use their
1416 own configuration files.
1418 * Added 'set_var' option for an endpoint. For each variable specified that
1419 variable gets set upon creation of a channel involving the endpoint.
1423 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1424 them, an Asterisk-specific version of PJSIP needs to be installed.
1425 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1427 res_statsd/res_chan_stats
1429 * A new resource module, res_statsd, has been added, which acts as a statsd
1430 client. This module allows Asterisk to publish statistics to a statsd
1431 server. In conjunction with res_chan_stats, it will publish statistics about
1432 channels to the statsd server. It can be configured via res_statsd.conf.
1436 * Device state for XMPP buddies is now available using the following format:
1437 XMPP/<client name>/<buddy address>
1438 If any resource is available the device state is considered to be not in use.
1439 If no resources exist or all are unavailable the device state is considered
1446 Realtime/Database Scripts
1448 * Asterisk previously included example db schemas in the contrib/realtime/
1449 directory of the source tree. This has been replaced by a set of database
1450 migrations using the Alembic framework. This allows you to use alembic to
1451 initialize the database for you. It will also serve as a database migration
1452 tool when upgrading Asterisk in the future.
1454 See contrib/ast-db-manage/README.md for more details.
1458 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1459 This python script will convert an existing sip.conf file to a
1460 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1461 is meant to be an aid in converting an existing chan_sip configuration to
1462 a chan_pjsip configuration, but it is expected that configuration beyond
1463 what the script provides will be needed.
1466 ------------------------------------------------------------------------------
1467 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1468 ------------------------------------------------------------------------------
1472 * The Asterisk build system will now build and install a shared library
1473 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1474 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1475 that Asterisk can ensure that these functions do *not* get called by any
1476 modules that are loaded into Asterisk, since they should only be called once
1477 in any single process. If desired, this feature can be disabled by supplying
1478 the "--disable-asteriskssl" option to the configure script.
1480 * A new make target, 'full', has been added to the Makefile. This performs
1481 the same compilation actions as make all, but will also scan the entirety of
1482 each source file for documentation. This option is needed to generate AMI
1483 event documentation. Note that your system must have Python in order for
1484 this make target to succeed.
1486 * The optimization portion of the build system has been reworked to avoid
1487 broken builds on certain architectures. All architecture-specific
1488 optimization has been removed in favor of using -march=native to allow gcc
1489 to detect the environment in which it is running when possible. This can
1490 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1492 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1493 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1495 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1496 previously parsed the header file to obtain the version of Asterisk, you
1497 will now have to go through Asterisk to get the version information.
1505 * Added 'F()' option. Similar to the dial option, this can be supplied with
1506 arguments indicating where the callee should go after the caller is hung up,
1507 or without options specified, the priority after the Queue will be used.
1512 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1513 all non-admin participants on a conference. The confbridge configuration
1514 file also allows for the default sounds played to all conference users when
1515 this occurs to be overriden using sound_participants_unmuted and
1516 sound_participants_muted.
1518 * Added menu action participant_count. This will playback the number of
1519 current participants in a conference.
1521 * Added announcement configuration option to user profile. If set the sound
1522 file will be played to the user, and only the user, upon joining the
1525 * Added record_file_append option that defaults to "yes", but if set to no
1526 will create a new file between each start/stop recording.
1531 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1532 channels respectively before the callee channels are called.
1537 * Added support for IPv6.
1539 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1540 external process will cause the current playlist to be cleared, including
1541 stopping any audio file that is currently playing. This is useful when you
1542 want to interrupt audio playback only when specific DTMF is entered by the
1548 * A new option, 'I' has been added to app_followme. By setting this option,
1549 Asterisk will not update the caller with connected line changes when they
1550 occur. This is similar to app_dial and app_queue.
1552 * The 'N' option is now ignored if the call is already answered.
1554 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1555 and caller channels respectively before the callee channels are called.
1557 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1563 * MixMonitor hooks now have IDs associated with them which can be used to
1564 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1565 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1566 now accepts that ID as an argument.
1568 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1569 indicated mailboxes.
1574 * The connect action in app_mysql now allows you to specify a port number to
1575 connect to. This is useful if you run a MySQL server on a non-standard
1581 * Increased the default number of allowed destinations from 5 to 12.
1586 * The app_page application now no longer depends on DAHDI or app_meetme. It
1587 has been re-architected to use app_confbridge internally.
1592 * Added queue options autopausebusy and autopauseunavail for automatically
1593 pausing a queue member when their device reports busy or congestion.
1595 * The 'ignorebusy' option for queue members has been deprecated in favor of
1596 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1597 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1598 per interface basis. Individual ringinuse values can now be set in
1599 queues.conf via an argument to member definitions. Lastly, the queue
1600 'ringinuse' setting now only determines defaults for the per member
1601 'ringinuse' setting and does not override per member settings like it does
1602 in earlier versions.
1604 * Added 'F()' option. Similar to the dial option, this can be supplied with
1605 arguments indicating where the callee should go after the caller is hung up,
1606 or without options specified, the priority after the Queue will be used.
1608 * Added new option log_member_name_as_agent, which will cause the membername to
1609 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1610 state_interface has been set.
1612 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1614 * App_queue will now play periodic announcements for the caller that
1615 holds the first position in the queue while waiting for answer.
1619 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1620 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1621 changed arguments to SayUnixTime so that every option is truly optional even
1622 when using multiple options (so that j option could be used without having to
1623 manually specify timezone and format) There are other benefits, e.g., format
1624 can now be used without specifying time zone as well.
1629 * Addition of the VM_INFO function - see Function changes.
1631 * The imapserver, imapport, and imapflags configuration options can now be
1632 overriden on a user by user basis.
1634 * When voicemail plays a message's envelope with saycid set to yes, when
1635 reaching the caller id field it will play a recording of a file with the same
1636 base name as the sender's callerid if there is a similarly named file in
1637 <astspooldir>/recordings/callerids/
1639 * Voicemails now contains a unique message identifier "msg_id", which is stored
1640 in the message envelope with the sound files. IMAP backends will now store
1641 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1642 backends will store the message identifier in a "msg_id" column. See
1643 UPGRADE.txt for more information.
1645 * Added VoiceMailPlayMsg application. This application will play a single
1646 voicemail message from a mailbox. The result of the application, SUCCESS or
1647 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1652 * Hangup handlers can be attached to channels using the CHANNEL() function.
1653 Hangup handlers will run when the channel is hung up similar to the h
1654 extension. The hangup_handler_push option will push a GoSub compatible
1655 location in the dialplan onto the channel's hangup handler stack. The
1656 hangup_handler_pop option will remove the last added location, and optionally
1657 replace it with a new GoSub compatible location. The hangup_handler_wipe
1658 option will remove all locations on the stack, and optionally add a new
1661 * The expression parser now recognizes the ABS() absolute value function,
1662 which will convert negative floating point values to positive values.
1664 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1665 control of faxdetect.
1667 * Addition of the VM_INFO function that can be used to retrieve voicemail
1668 user information, such as the email address and full name.
1669 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1672 * The REDIRECTING function now supports the redirecting original party id
1675 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1676 lets you set some of the configuration options from the [general] section
1677 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1678 the key sequence used to activate built-in features, such as blindxfer,
1679 and automon. See the built-in documentation for details.
1681 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1682 instead of simply the uri. This is the format that MessageSend() can use
1683 in the from parameter for outgoing SIP messages.
1685 * Added the PRESENCE_STATE function. This allows retrieving presence state
1686 information from any presence state provider. It also allows setting
1687 presence state information from a CustomPresence presence state provider.
1688 See AMI/CLI changes for related commands.
1690 * Added the AMI_CLIENT function to make manager account attributes available
1691 to the dialplan. It currently supports returning the current number of
1692 active sessions for a given account.
1694 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1695 and the REDIRECTING functions.
1703 * Added a manager event "LocalBridge" for local channel call bridges between
1704 the two pseudo-channels created.
1709 * Added dialtone_detect option for analog ports to disconnect incoming
1710 calls when dialtone is detected.
1712 * Added option colp_send to send ISDN connected line information. Allowed
1713 settings are block, to not send any connected line information; connect, to
1714 send connected line information on initial connect; and update, to send
1715 information on any update during a call. Default is update.
1717 * Add options namedcallgroup and namedpickupgroup to support installations
1718 where a higher number of groups (>64) is required.
1720 * Added support to use private party ID information with PRI calls.
1725 * A new channel driver named chan_motif has been added which provides support for
1726 Google Talk and Jingle in a single channel driver. This new channel driver includes
1727 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1728 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1729 specification, current Google Jingle specification, and the original Google Talk
1735 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1736 globally and overriden on a peer by peer basis.
1738 * Direct media functionality has been added. Options in config are:
1739 directmedia (directrtp) and directrtpsetup (earlydirect)
1741 * ChannelUpdate events now contain a CallRef header.
1746 * Asterisk will no longer substitute CID number for CID name in the display
1747 name field if CID number exists without a CID name. This change improves
1748 compatibility with certain device features such as Avaya IP500's directory
1751 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1752 created using that setting to not be removed during SIP reload.
1754 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1755 request with a "Record:" header, this will turn the requested feature on/off.
1756 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1757 dynamic features must be enabled and configured properly on the requesting
1758 channel for this to function properly.
1760 * Add support to realtime for the 'callbackextension' option.
1762 * When multiple peers exist with the same address, but differing
1763 callbackextension options, incoming requests that are matched by address
1764 will be matched to the peer with the matching callbackextension if it is
1767 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1768 which set the force_rport and comedia options automatically if Asterisk
1769 detects that an incoming SIP request crossed a NAT after being sent by
1770 the remote endpoint.
1772 * The default global nat setting in sip.conf has been changed from force_rport
1773 to auto_force_rport.
1775 * NAT settings are now a combinable list of options. The equivalent of the
1776 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1778 * Adds an option send_diversion which can be disabled to prevent
1779 diversion headers from automatically being added to INVITE requests.
1781 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1782 be sent to the remote host at a given interval to keep the NAT mapping open.
1783 This can be enabled using the keepalive configuration option.
1785 * Add option 'tonezone' to specify country code for indications. This option
1786 can be set both globally and overridden for specific peers.
1788 * The SIP Security Events Framework now supports IPv6.
1790 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1791 between multiple user agents. When set, for directmedia reinvites,
1792 Asterisk will not send an immediate reinvite on an incoming call leg. This
1793 option is useful when peered with another SIP user agent that is known to
1794 send immediate direct media reinvites upon call establishment.
1796 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1799 * Add options subminexpiry and submaxexpiry to set limits of subscription
1800 timer independently from registration timer settings. The setting of the
1801 registration timer limits still is done by options minexpiry, maxexpiry
1802 and defaultexpiry. For backwards compatibility the setting of minexpiry
1803 and maxexpiry also is used to configure the subscription timer limits if
1804 subminexpiry and submaxexpiry are not set in sip.conf.
1806 * Set registration timer limits to default values when reloading sip
1807 configuration and values are not set by configuration.
1809 * Add options namedcallgroup and namedpickupgroup to support installations
1810 where a higher number of groups (>64) is required.
1812 * When a MESSAGE request is received, the address the request was received from
1813 is now saved in the SIP_RECVADDR variable.
1815 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1816 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1817 the ANI2/OLI information is set on the channel, which can be retrieved using
1818 the CALLERID function.
1820 * Peers can now be configured to support negotiation of ICE candidates using
1821 the setting icesupport. See res_rtp_asterisk changes for more information.
1823 * Added support for format attribute negotiation. See the Codecs changes for
1826 * Extra headers specified with SIPAddHeader are sent with the REFER message
1827 when using Transfer application. See refer_addheaders in sip.conf.sample.
1829 * Added support to use private party ID information with calls.
1831 * Adds an option discard_remote_hold_retrieval that when set stops telling
1832 the peer to start music on hold.
1837 * Added skinny version 17 protocol support.
1841 --------------------
1842 * Added ability to use multiple lines for a single phone. This allows multiple
1843 calls to occur on a single phone, using callwaiting and switching between calls.
1845 * Added option 'sharpdial' allowing end dialing by pressing # key
1847 * Added option 'interdigit_timer' to control phone dial timeout
1849 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1851 * Added global 'debug' option, that enables debug in channel driver
1853 * Added ability to translate on-screen menu in multiple languages. Tested on
1854 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1855 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1858 * In addition to English added French and Russian languages for on-screen menus
1860 * Reworked dialing number input: added dialing by timeout, immediate dial on
1861 on dialplan compare, phone number length now not limited by screen size
1863 * Added ability to pickup a call using features.conf defined value and
1869 * Add options namedcallgroup and namedpickupgroup to support installations
1870 where a higher number of groups (>64) is required.
1872 * Added support to use private party ID information with calls.
1877 * The minimum DTMF duration can now be configured in asterisk.conf
1878 as "mindtmfduration". The default value is (as before) set to 80 ms.
1879 (previously it was only available in source code)
1881 * Named ACLs can now be specified in acl.conf and used in configurations that
1882 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1883 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1884 working ACL. In addition, some CLI commands have been added to provide
1885 show information and allow for module reloading - see CLI Changes.
1887 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1888 items (separated by commas), and items in the rule can be negated by prefixing
1889 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1890 longer necessray to control the order that the 'permit' and 'deny' columns are
1891 returned from queries.
1893 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1894 be used within the dynamic weight attribute when specifying a mapping.
1896 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1897 header, instead of putting the user defined event name there. When enabled
1898 the UserDefType header is added for user defined events. This feature is
1899 enabled with the setting show_user_defined.
1901 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1902 line purposes use the following variables instead of their macro equivalents:
1903 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1904 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1905 cc_callback_macro in channel configurations.
1907 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1910 * Call files now support the "early_media" option to connect with an outgoing
1911 extension when early media is received.
1913 * Added support to use private party ID information with calls.
1918 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1919 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1920 AGI application would exit immediately after a channel hangup is detected.
1922 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1923 are resolved and each address is attempted in turn until one succeeds or
1927 AMI (Asterisk Manager Interface)
1929 * The originate action now has an option "EarlyMedia" that enables the
1930 call to bridge when we get early media in the call. Previously,
1931 early media was disregarded always when originating calls using AMI.
1933 * Added setvar= option to manager accounts (much like sip.conf)
1935 * Originate now generates an error response if the extension given is not found
1938 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1939 them if the i(variable) option is used. StopMixMonitor will accept
1940 MixMonitorID as an option to close specific MixMonitors.
1942 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1943 updated to include information about peers configured with
1944 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1945 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1946 returned if auto_force_rport is not enabled.
1948 * Added SIPpeerstatus manager command which will generate PeerStatus events
1949 similar to the existing PeerStatus events found in chan_sip on demand.
1951 * Hangup now can take a regular expression as the Channel option. If you want
1952 to hangup multiple channels, use /regex/ as the Channel option. Existing
1953 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1954 the manager will send you a list of channels back that were hung up.
1956 * Support for IPv6 addresses has been added.
1958 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1959 documentation is only generated when Asterisk is compiled using 'make full'.
1960 See the CLI section for commands to display AMI event information.
1962 * The AMI Hangup event now includes the AccountCode header so you can easily
1963 correlate with AMI Newchannel events.
1965 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1966 the StateInterface of the queue member.
1968 * Added AMI event SessionTimeout in the Call category that is issued when a
1969 call is terminated due to either RTP stream inactivity or SIP session timer
1972 * CEL events can now contain a user defined header UserDefType. See core
1973 changes for more information.
1975 * OOH323 ChannelUpdate events now contain a CallRef header.
1977 * Added PresenceState command. This command will report the presence state for
1978 the given presence provider.
1980 * Added Parkinglots command. This will list all parking lots as a series of
1981 AMI Parkinglot events.
1983 * Added MessageSend command. This behaves in the same manner as the
1984 MessageSend application, and is a technolgoy agnostic mechanism to send out
1985 of call text messages.
1987 * Added "message" class authorization. This grants an account permission to
1988 send out of call messages. Write-only.
1993 * The "dialplan add include" command has been modified to create context a context
1994 if one does not already exist. For instance, "dialplan add include foo into bar"
1995 will create context "bar" if it does not already exist.
1997 * A "dialplan remove context" command has been added to remove a context from
2000 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2001 filenames of all running mixmonitors on a channel.
2003 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2004 numeric instead of 0, 1, or 2.
2006 * "stun show status" will show a table describing how the STUN client is
2009 * "acl show [named acl]" will show information regarding a Named ACL. The
2010 acl module can be reloaded with "reload acl".
2012 * Added CLI command to display AMI event information - "manager show events",
2013 which shows a list of all known and documented AMI events, and "manager show
2014 event [event name]", which shows detail information about a specific AMI
2017 * The result of the CLI command "queue show" now includes the state interface
2018 information of the queue member.
2020 * The command "core set verbose" will now set a separate level of logging for
2021 each remote console without affecting any other console.
2023 * Added command "cdr show pgsql status" to check connection status
2025 * "sip show channel" will now display the complete route set.
2027 * Added "presencestate list" command. This command will list all custom
2028 presence states that have been set by using the PRESENCE_STATE dialplan
2031 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2032 command. This changes a custom presence to a new state.
2037 * Codec lists may now be modified by the '!' character, to allow succinct
2038 specification of a list of codecs allowed and disallowed, without the
2039 requirement to use two different keywords. For example, to specify all
2040 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2042 * Add support for parsing SDP attributes, generating SDP attributes, and
2043 passing it through. This support includes codecs such as H.263, H.264, SILK,
2044 and CELT. You are able to set up a call and have attribute information pass.
2045 This should help considerably with video calls.
2047 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2048 just like the GSM codec.
2052 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2053 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2057 * Asterisk version and build information is now logged at the beginning of a
2060 * Threads belonging to a particular call are now linked with callids which get
2061 added to any log messages produced by those threads. Log messages can now be
2062 easily identified as involved with a certain call by looking at their call id.
2063 Call ids may also be attached to log messages for just about any case where
2064 it can be determined to be related to a particular call.
2066 * Each logging destination and console now have an independent notion of the
2067 current verbosity level. Logger.conf now allows an optional argument to
2068 the 'verbose' specifier, indicating the level of verbosity sent to that
2069 particular logging destination. Additionally, remote consoles now each
2070 have their own verbosity level. The command 'core set verbose' will now set
2071 a separate level for each remote console without affecting any other
2077 * Added 'announcement' option which will play at the start of MOH and between
2078 songs in modes of MOH that can detect transitions between songs (eg.
2084 * New per parking lot options: comebackcontext and comebackdialtime. See
2085 configs/features.conf.sample for more details.
2087 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2090 * Channel variable PARKEDCALL is now set with the name of the parking lot
2091 when a timeout occurs.
2097 CDR Postgresql Driver
2099 * Added command "cdr show pgsql status" to check connection status
2102 CDR Adaptive ODBC Driver
2104 * Added schema option for databases that support specifying a schema.
2112 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2113 CALENDAR_WRITE has completed successfully.
2118 * A new option, 'probation' has been added to rtp.conf
2119 RTP in strictrtp mode can now require more than 1 packet to exit learning
2120 mode with a new source (and by default requires 4). The probation option
2121 allows the user to change the required number of packets in sequence to any
2122 desired value. Use a value of 1 to essentially restore the old behavior.
2123 Also, with strictrtp on, Asterisk will now drop all packets until learning
2124 mode has successfully exited. These changes are based on how pjmedia handles
2125 media sources and source changes.
2127 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2128 enabled or disabled using the icesupport setting. A variety of other
2129 settings have been introduced to configure STUN/TURN connections.
2134 * A new module, res_corosync, has been introduced. This module uses the
2135 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2136 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2137 Device State (presence) information. This module is very similar to, and
2138 is a replacement for the res_ais module that was in previous releases of
2144 * This module adds a cleaned up, drop-in replacement for res_jabber called
2145 res_xmpp. This provides the same externally facing functionality but is
2146 implemented differently internally. res_jabber has been deprecated in favor
2147 of res_xmpp; please see the UPGRADE.txt file for more information.
2152 * The safe_asterisk script has been updated to allow several of its parameters
2153 to be set from environment variables. This also enables a custom run
2154 directory of Asterisk to be specified, instead of defaulting to /tmp.
2156 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2157 its value to determine the directory to assume is the top-level directory of
2158 the source tree. If the variable is not set, it defaults to the current
2159 behavior and uses the current working directory.
2161 ------------------------------------------------------------------------------
2162 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2163 ------------------------------------------------------------------------------
2167 * Asterisk now has protocol independent support for processing text messages
2168 outside of a call. Messages are routed through the Asterisk dialplan.
2169 SIP MESSAGE and XMPP are currently supported. There are options in
2170 jabber.conf and sip.conf to allow enabling these features.
2171 -> jabber.conf: see the "sendtodialplan" and "context" options.
2172 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2173 and "outofcall_message_context" options.
2174 The MESSAGE() dialplan function and MessageSend() application have been
2175 added to go along with this functionality. More detailed usage information
2176 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2177 * If real-time text support (T.140) is negotiated, it will be preferred for
2178 sending text via the SendText application. For example, via SIP, messages
2179 that were once sent via the SIP MESSAGE request would be sent via RTP if
2180 T.140 text is negotiated for a call.
2184 * parkedmusicclass can now be set for non-default parking lots.
2186 Asterisk Manager Interface
2187 --------------------------
2188 * PeerStatus now includes Address and Port.
2189 * Added Hold events for when the remote party puts the call on and off hold
2190 for chan_dahdi ISDN channels.
2191 * Added new action MeetmeListRooms to list active conferences (shows same
2192 data as "meetme list" at the CLI).
2193 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2194 Description field that is set by 'description' in the channel configuration
2196 * Added Uniqueid header to UserEvent.
2197 * Added new action FilterAdd to control event filters for the current session.
2198 This requires the system permission and uses the same filter syntax as
2199 filters that can be defined in manager.conf
2200 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2201 versions had some instances of the event converted, but others were left
2202 as-is. All Unlink events should now be converted to Bridge events. The AMI
2203 protocol version number was incremented to 1.2 as a result of this change.
2205 Asterisk HTTP Server
2206 --------------------------
2207 * The HTTP Server can bind to IPv6 addresses.
2210 --------------------------
2211 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2212 with busydetect. usage example: busypattern=200,200,200,600
2215 --------------------------
2216 * New 'gtalk show settings' command showing the current settings loaded from
2218 * The 'logger reload' command now supports an optional argument, specifying an
2219 alternate configuration file to use.
2220 * 'dialplan add extension' command will now automatically create a context if
2221 the specified context does not exist with a message indicated it did so.
2222 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2223 Description field which can be populated with 'description' in the channel
2224 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2227 --------------------------
2228 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2229 thus allowing records which do NOT match the specified filter.
2230 * Added ability to log CONGESTION calls to CDR
2233 --------------------------
2234 * Ability to define custom SILK formats in codecs.conf.
2235 * Addition of speex32 audio format with translation.
2236 * CELT codec pass-through support and ability to define
2237 custom CELT formats in codecs.conf.
2238 * Ability to read raw signed linear files with sample rates
2239 ranging from 8khz - 192khz. The new file extensions introduced
2240 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2241 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2242 Skinny, H.323, etc) can still only support the following codecs:
2243 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2244 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2245 Video: h261, h263, h263p, h264, mpeg4
2250 --------------------------
2251 * New highly optimized and customizable ConfBridge application capable of
2252 mixing audio at sample rates ranging from 8khz-96khz.
2253 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2254 and bridge profiles on a channel.
2255 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2256 about a conference such as locked status and number of parties, admins,
2258 * Addition of video_mode option in confbridge.conf for adding video support
2259 into a bridge profile.
2260 * Addition of the follow_talker video_mode in confbridge.conf. This video
2261 mode dynamically switches the video feed to always display the loudest talker
2262 supplying video in the conference.
2266 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2267 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2268 variables from asterisk.conf.
2272 * Addition of the JITTERBUFFER dialplan function. This function allows
2273 for jitterbuffering to occur on the read side of a channel. By using
2274 this function conference applications such as ConfBridge and MeetMe can
2275 have the rx streams jitterbuffered before conference mixing occurs.
2276 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2278 * Added STRREPLACE function. This function let's the user search a variable
2279 for a given string to replace with another string as many times as the
2280 user specifies or just throughout the whole string.
2281 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2282 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2283 * Added extensions to chan_ooh323 in function CHANNEL()
2285 libpri channel driver (chan_dahdi) DAHDI changes
2286 --------------------------
2287 * Added moh_signaling option to specify what to do when the channel's bridged
2288 peer puts the ISDN channel on hold.
2289 * Added display_send and display_receive options to control how the display ie
2290 is handled. To send display text from the dialplan use the SendText()
2291 application when the option is enabled.
2292 * Added mcid_send option to allow sending a MCID request on a span.
2295 --------------------------
2296 * Added setvar option to calendar.conf to allow setting channel variables on
2297 notification channels.
2298 * Added "calendar show types" CLI command to list registered calendar
2302 --------------------------
2303 * Added two new options, r and t with file name arguments to record
2304 single direction (unmixed) audio recording separate from the bidirectional
2305 (mixed) recording. The mixed file name argument is optional now as long
2306 as at least one recording option is used.
2309 --------------------------
2310 * Added a new option, l, which will disable local call optimization for
2311 channels involved with the FollowMe thread. Use this option to improve
2312 compatability for a FollowMe call with certain dialplan apps, options, and
2316 --------------------------
2317 * Added option "k" that will automatically close the conference when there's
2318 only one person left when a user exits the conference.
2321 --------------------------
2322 * cel_pgsql now supports the 'extra' column for data added using the
2323 CELGenUserEvent() application.
2326 --------------------------
2327 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2328 in the sample extensions.lua file for syntax details.
2329 * Applications that perform jumps in the dialplan such as Goto will now
2330 execute properly. When pbx_lua detects that the context, extension, or
2331 priority we are executing on has changed it will immediately return control
2332 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2333 the priority after the currently executing priority.
2334 * An autoservice is now started by default for pbx_lua channels. It can be
2335 stopped and restarted using the autoservice_stop() and autoservice_start()
2339 --------------------------
2340 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2341 into a FAXStatus event with an 'Operation' header that will be either
2342 'send', 'receive', and 'gateway'.
2343 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2344 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2345 feature will handle converting a fax call between an audio T.30 fax terminal
2346 and an IFP T.38 fax terminal.
2350 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2351 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2352 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2356 * Added general option negative_penalty_invalid default off. when set
2357 members are seen as invalid/logged out when there penalty is negative.
2358 for realtime members when set remove from queue will set penalty to -1.
2359 * Added queue option autopausedelay when autopause is enabled it will be
2360 delayed for this number of seconds since last successful call if there
2361 was no prior call the agent will be autopaused immediately.
2362 * Added member option ignorebusy this when set and ringinuse is not
2363 will allow per member control of multiple calls as ringinuse does for
2368 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2370 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2371 one participant left (much like a normal call bridge)
2372 * Added extra argument to Originate to set timeout.
2376 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2377 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2378 utility in the UTILS section of menuselect. If an existing astdb is found and no
2379 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2380 convert an existing astdb to the SQLite3 version automatically at runtime.
2384 * Modules marked as deprecated are no longer marked as building by default. Enabling
2385 these modules is still available via menuselect.
2389 * authdebug is now disabled by default. To enable this functionaility again
2390 set authdebug = yes in iax.conf.
2394 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2395 releases it was disabled.
2399 * The PBX core previously made a call with a non-existing extension test for
2400 extension s@default and jump there if the extension existed.
2401 This was a bad default behaviour and violated the principle of least surprise.
2402 It has therefore been changed in this release. It may affect some
2403 applications and configurations that rely on this behaviour. Most channel
2404 drivers have avoided this for many releases by testing whether the extension
2405 called exists before starting the PBX and generating a local error.
2406 This behaviour still exists and works as before.
2408 Extension "s" is used when no extension is given in a channel driver,
2409 like immediate answer in DAHDI or calling to a domain with no user part
2412 ------------------------------------------------------------------------------
2413 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2414 ------------------------------------------------------------------------------
2418 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2419 now defaults to force_rport. It is very important that phones requiring nat=no be
2420 specifically set as such instead of relying on the default setting. If at all
2421 possible, all devices should have nat settings configured in the general section as
2422 opposed to configuring nat per-device.
2423 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2424 codecs sent in response to an INVITE to the single most preferred codec.
2425 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2426 to be used for the outgoing call. It must be one of the codecs configured
2428 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2429 to be used for holding a private key. If tlsprivatekey is not specified,
2430 tlscertfile is searched for both public and private key.
2431 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2432 outbound client connections to be specified.
2433 * The sendrpid parameter has been expanded to include the options
2434 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2435 header to be sent (equivalent to setting sendrpid=yes) and setting
2436 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2437 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2438 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2439 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2440 will accept the SDP even if the SDP version number is not properly incremented,
2441 but will generate a warning in the log indicating that the SIP peer that sent
2442 the SDP should have the 'ignoresdpversion' option set.
2443 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2444 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2445 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2446 remote side requests it and disables symmetric RTP support. Setting it to
2447 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2448 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2449 and enables symmetric RTP support.
2450 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2451 response. This permits the master channel to know how each channel dialled
2452 in a multi-channel setup resolved in an individual way. This carries a
2453 performance penalty and can be disabled in sip.conf using the
2454 'storesipcause' option.
2455 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2456 configuration for the externip and externhost options when tcp or tls is used.
2457 * Added support for message body (stored in content variable) to SIP NOTIFY message
2458 accessible via AMI and CLI.
2459 * Added 'media_address' configuration option which can be used to explicitly specify
2460 the IP address to use in the SDP for media (audio, video, and text) streams.
2461 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2462 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2464 * Added 'use_q850_reason' configuration option for generating and parsing
2465 if available Reason: Q.850;cause=<cause code> header. It is implemented
2466 in some gateways for better passing PRI/SS7 cause codes via SIP.
2467 * When dialing SIP peers, a new component may be added to the end of the dialstring
2468 to indicate that a specific remote IP address or host should be used when dialing
2469 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2470 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2471 ability to selectively force bridged channels to also be encrypted is also
2472 implemented. Branching in the dialplan can be done based on whether or not
2473 a channel has secure media and/or signaling.
2474 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2476 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2477 Charge messages to snom phones.
2478 * Added support for G.719 media streams.
2479 * Added support for 16khz signed linear media streams.
2480 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2481 RTP has been outfitted with the same abilities.
2482 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2483 available in device configurations as well as in the dial plan.
2484 * Addition of the 'subscribe_network_change' option for turning on and off
2485 res_stun_monitor module support in chan_sip.
2486 * Addition of the 'auth_options_requests' option for turning on and off
2487 authentication for OPTIONS requests in chan_sip.
2491 * Add #tryinclude statement for config files. This provides the same
2492 functionality as the #include statement however an asterisk module will
2493 still load if the filename does not exist. Using the #include statement
2494 Asterisk will not allow the module to load.
2498 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2499 on realtime updates.
2500 * Added the ability for chan_iax2 to inform the dialplan whether or not
2501 encryption is being used. This interoperates with the SIP SRTP implementation
2502 so that a secure SIP call can be bridged to a secure IAX call when the
2503 dialplan requires bridged channels to be "secure".
2504 * Addition of the 'subscribe_network_change' option for turning on and off
2505 res_stun_monitor module support in chan_iax.
2510 * Added ability to preset channel variables on indicated lines with the setvar
2511 configuration option. Also, clearvars=all resets the list of variables back
2513 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2514 See configs/res_pktccops.conf for more information.
2516 XMPP Google Talk/Jingle changes
2517 -------------------------------
2518 * Added the externip option to gtalk.conf.
2519 * Added the stunaddr option to gtalk.conf which allows for the automatic
2520 retrieval of the external ip from a stun server.
2524 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2525 match to a partial channel name.
2526 * Added .m3u support for Mp3Player application.
2527 * Added progress option to the app_dial D() option. When progress DTMF is
2528 present, those values are sent immediately upon receiving a PROGRESS message
2529 regardless if the call has been answered or not.
2530 * Added functionality to the app_dial F() option to continue with execution
2531 at the current location when no parameters are provided.
2532 * Added the 'a' option to app_dial to answer the calling channel before any
2533 announcements or macros are executed.
2534 * Modified app_dial to set answertime when the called channel answers even if
2535 the called channel hangs up during playback of an announcement.
2536 * Modified app_dial 'r' option to support an additional parameter to play an
2537 indication tone from indications.conf
2538 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2539 to cycle through the next available channel. By default this is still '*'.
2540 * Added x() option to app_chanspy. This option allows DTMF to be set to
2541 exit the application.
2542 * The Voicemail application has been improved to automatically ignore messages
2543 that only contain silence.
2544 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2545 associated mailbox(es) to be greetings-only.
2546 * The ChanSpy application now has the 'S' option, which makes the application
2547 automatically exit once it hits a point where no more channels are available
2549 * The ChanSpy application also now has the 'E' option, which spies on a single
2550 channel and exits when that channel hangs up.
2551 * The MeetMe application now turns on the DENOISE() function by default, for
2552 each participant. In our tests, this has significantly decreased background
2553 noise (especially noisy data centers).
2554 * Voicemail now permits storage of secrets in a separate file, located in the
2555 spool directory of each individual user. The control for this is located in
2556 the "passwordlocation" option in voicemail.conf. Please see the sample
2557 configuration for more information.
2558 * The ChanIsAvail application now exposes the returned cause code using a separate
2559 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2560 * Added 'd' option to app_followme. This option disables the "Please hold"
2562 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2563 received will terminate recording.
2564 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2565 Previously the folder could only be set per context, but has now been extended
2566 using the imapfolder option.
2567 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2568 * Voicemail now allows the pager date format to be specified separately from the
2570 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2571 to allow joining, leaving, and sending text to group chats.
2572 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2573 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2574 to all paged phones (and optionally excluding the caller's one using the new
2575 option 'n') before the call is bridged.
2576 * The 'f' option to Dial has been augmented to take an optional argument. If no
2577 argument is provided, the 'f' option works as it always has. If an argument is
2578 provided, then the connected party information of all outgoing channels created
2579 during the Dial will be set to the argument passed to the 'f' option.
2580 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2582 * The OSP lookup application adds in/outbound network ID, optional security,
2583 number portability, QoS reporting, destination IP port, custom info and service
2585 * Added new application VMSayName that will play the recorded name of the voicemail
2586 user if it exists, otherwise will play the mailbox number.
2587 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2588 retrieve state for a particular bridge, where <name> is the conference name
2589 * app_directory now allows exiting at any time using the operator or pound key.
2590 * Voicemail now supports setting a locale per-mailbox.
2591 * Two new applications are provided for declining counting phrases in multiple
2592 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2594 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2596 * Voicemail now includes rdnis within msgXXXX.txt file.
2597 * ExternalIVR now supports IPv6 addresses.
2598 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2599 at https://wiki.asterisk.org/wiki/x/oQBB
2600 * ParkedCall and Park can now specify the parking lot to use.
2604 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2605 over SRV records associated with a specific service. From the CLI, type
2606 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2607 details on how these may be used.
2608 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2609 pitch of a channel's tx and rx audio streams.
2610 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2611 setting various connected line and redirecting party information.
2612 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2613 support ISDN subaddressing.
2614 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2615 * For DAHDI channels, the CHANNEL() dialplan function now allows
2616 the dialplan to request changes in the configuration of the active
2617 echo canceller on the channel (if any), for the current call only.
2620 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2622 The possible values are:
2624 on - normal mode (the echo canceller is actually reinitialized)
2626 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2628 voice - voice mode (returns from FAX mode, reverting the changes that
2629 were made when FAX mode was requested)
2630 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2631 and setting variables on the channel which created the current channel.
2632 Administrators should take care to avoid naming conflicts, when multiple
2633 channels are dialled at once, especially when used with the Local channel
2634 construct (which all could set variables on the master channel). Usage
2635 of the HASH() dialplan function, with the key set to the name of the slave
2636 channel, is one approach that will avoid conflicts.
2637 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2639 * func_odbc now allows multiple row results to be retrieved without using
2640 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2641 from the same query by using the name of the function which retrieved the
2642 first row as an argument to ODBC_FETCH().
2643 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2644 dialplan. This function returns the content of the received message.
2645 * Added REPLACE, which searches a given variable name for a set of characters,
2646 then either replaces them with a single character or deletes them.
2647 * Added PASSTHRU, which literally passes the same argument back as its return
2648 value. The intent is to be able to use a literal string argument to
2649 functions that currently require a variable name as an argument.
2650 * HASH-associated variables now can be inherited across channel creation, by
2651 prefixing the name of the hash at assignment with the appropriate number of
2652 underscores, just like variables.
2653 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2654 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2655 whether or not channels that are bridged to the current channel will be
2656 required to have secure signaling and/or media.
2657 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2658 the current channel has secure signaling and/or media.
2659 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2660 "no_media_path" option.
2661 Returns "0" if there is a B channel associated with the call.
2662 Returns "1" if no B channel is associated with the call. The call is either
2663 on hold or is a call waiting call.
2664 * Added option to dialplan function CDR(), the 'f' option
2665 allows for high resolution times for billsec and duration fields.
2666 * FILE() now supports line-mode and writing.
2667 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2668 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2672 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2673 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2674 and is set when a dynamic feature is triggered.
2675 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2676 to dynamically create a new parking lot matching the value this varible is
2678 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2679 features.conf that should be the base for dynamic parkinglots.
2680 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2681 parkinglot should have.
2682 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2683 parkinglot should have.
2684 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2689 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2690 timeout has expired.
2691 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2692 to the caller when an Agent's phone is ringing. This can be used to indicate
2693 to the caller that their call is about to be picked up, which is nice when
2694 one has been on hold for an extened period of time.
2695 * A new config option, penaltymemberslimit, has been added to queues.conf.
2696 When set this option will disregard penalty settings when a queue has too
2698 * A new option, 'I' has been added to both app_queue and app_dial.
2699 By setting this option, Asterisk will not update the caller with
2700 connected line changes or redirecting party changes when they occur.
2701 * A 'relative-periodic-announce' option has been added to queues.conf. When
2702 enabled, this option will cause periodic announce times to be calculated
2703 from the end of announcements rather than from the beginning.
2704 * The autopause option in queues.conf can be passed a new value, "all." The
2705 result is that if a member becomes auto-paused, he will be paused in all
2706 queues for which he is a member, not just the queue that failed to reach
2708 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2709 * The queue logger now allows events to optionally propagate to a file,
2710 even when realtime logging is turned on. Additionally, realtime logging
2711 supports sending the event arguments to 5 individual fields, although it
2712 will fallback to the previous data definition, if the new table layout is
2715 mISDN channel driver (chan_misdn) changes
2716 ----------------------------------------
2717 * Added display_connected parameter to misdn.conf to put a display string
2718 in the CONNECT message containing the connected name and/or number if
2719 the presentation setting permits it.
2720 * Added display_setup parameter to misdn.conf to put a display string
2721 in the SETUP message containing the caller name and/or number if the
2722 presentation setting permits it.
2723 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2724 indicate the dialplan settings are to be obtained from the asterisk
2726 * Made misdn.conf parameter callerid accept the "name" <number> format
2727 used by the rest of the system.
2728 * Made use the nationalprefix and internationalprefix misdn.conf
2729 parameters to prefix any received number from the ISDN link if that
2730 number has the corresponding Type-Of-Number. NOTE: This includes
2731 comparing the incoming call's dialed number against the MSN list.
2732 * Added the following new parameters: unknownprefix, netspecificprefix,
2733 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2734 received number from the ISDN link if that number has the corresponding
2736 * Added new dialplan application misdn_command which permits controlling
2737 the CCBS/CCNR functionality.
2738 * Added new dialplan function mISDN_CC which permits retrieval of various
2739 values from an active call completion record.
2740 * For PTP, you should manually send the COLR of the redirected-to party
2741 for an incomming redirected call if the incoming call could experience
2742 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2743 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2744 if the REDIRECTING(from-num) is not empty.
2745 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2746 option on all of the REDIRECTING statements before dialing the
2747 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2748 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2749 redirecting-to presentation (COLR) when it becomes available.
2750 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2753 thirdparty mISDN enhancements
2754 -----------------------------
2755 mISDN has been modified by Digium, Inc. to greatly expand facility message
2757 * Enhanced COLP support for call diversion and transfer.
2758 * CCBS/CCNR support.
2760 The latest modified mISDN v1.1.x based version is available at:
2761 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2762 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2764 Tagged versions of the modified mISDN code are available under:
2765 http://svn.digium.com/svn/thirdparty/mISDN/tags
2766 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2768 libpri channel driver (chan_dahdi) DAHDI changes
2769 -------------------------------------------
2770 * The channel variable PRIREDIRECTREASON is now just a status variable
2771 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2772 to read and alter the reason.
2773 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2774 redirected-to party for an incomming redirected call if the incoming call
2775 could experience further redirects. Just set the
2776 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2777 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2779 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2780 use the inhibit(i) option on all of the REDIRECTING statements before
2781 dialing the redirected-to party. You still have to set the
2782 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2783 will update the redirecting-to presentation (COLR) when it becomes available.
2784 * Added the ability to ignore calls that are not in a Multiple Subscriber
2785 Number (MSN) list for PTMP CPE interfaces.
2786 * Added dynamic range compression support for dahdi channels. It is
2787 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2788 * Added support for ISDN calling and called subaddress with partial support
2789 for connected line subaddress.
2790 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2791 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2792 to transfer a held call on disconnect similar to an analog phone.
2793 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2794 Will reroute/deflect an outgoing call when receive the message.
2795 Can use the DAHDISendCallreroutingFacility to send the message for the
2797 * Added standard location to add options to chan_dahdi dialing:
2798 Dial(DAHDI/g1[/extension[/options]])
2801 R Reverse charging indication
2802 * Added Reverse Charging Indication (Collect calls) send/receive option.
2803 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2804 Dial(DAHDI/g1/extension/R)
2805 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2806 (requires latest LibPRI)
2807 * Added ability to send/receive keypad digits in the SETUP message.
2808 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2809 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2810 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2811 (requires latest LibPRI)
2812 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2813 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2814 back into the same interface. Tromboned calls happen because of call routing,
2815 call deflection, call forwarding, and call transfer.
2816 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2817 * Added the ability to support call waiting calls. (The SETUP has no B channel
2819 * Added Malicious Call ID (MCID) event to the AMI call event class.
2820 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2822 Asterisk Manager Interface
2823 --------------------------
2824 * The Hangup action now accepts a Cause header which may be used to
2825 set the channel's hangup cause.
2826 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2827 to specify a separate .pem file to hold a private key. By default sslcert
2828 is used to hold both the public and private key.
2829 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2830 for options containing the 'tls' prefix. For example, 'sslenable' is now
2831 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2832 across all .conf files. All affected sample.conf files have been modified to
2833 reflect this change. Previous options such as 'sslenable' still work,
2834 but options with the 'tls' prefix are preferred.
2835 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2836 in a channel. (res_mutestream.so)
2837 * The configuration file manager.conf now supports a channelvars option, which
2838 specifies a list of channel variables to include in each channel-oriented
2840 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2841 and ExtraPriority to allow redirecting the second channel to a different
2842 location than the first.
2843 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2845 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2846 in a MixMonitor recording.
2847 * The 'iax2 show peers' output is now similar to the expected output of
2849 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2851 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2852 AOC-E messages on a channel.
2853 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2854 conform more closely to similar events.
2855 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2857 * Added optional parkinglot variable for park command.
2858 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2859 if CallerIDNum and CallerIDName headers are also present.
2861 Channel Event Logging
2862 ---------------------
2863 * A new interface, CEL, is introduced here. CEL logs single events, much like
2864 the AMI, but it differs from the AMI in that it logs to db backends much
2865 like CDR does; is based on the event subsystem introduced by Russell, and
2866 can share in all its benefits; allows multiple backends to operate like CDR;
2867 is specialized to event data that would be of concern to billing sytems,
2868 like CDR. Backends for logging and accounting calls have been produced,
2869 but a new CDR backend is still in development.
2873 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2874 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2875 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2876 * Multiple files and formats can now be specified in cdr_custom.conf.
2877 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2878 See configs/cdr_syslog.conf.sample for more information.
2879 * A 'sequence' field has been added to CDRs which can be combined with
2880 linkedid or uniqueid to uniquely identify a CDR.
2881 * Handling of billsec and duration field has changed. If your table definition
2882 specifies those fields as float,double or similar they will now be logged with
2883 microsecond accuracy instead of a whole integer.
2885 Calendaring for Asterisk
2886 ------------------------
2887 * A new set of modules were added supporing calendar integration with Asterisk.
2888 Dialplan functions for reading from and writing to calendars are included,
2889 as well as the ability to execute dialplan logic upon calendar event notifications.
2890 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2891 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2892 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2893 2003 support does not support forms-based authentication).
2895 Call Completion Supplementary Services for Asterisk
2896 ---------------------------------------------------
2897 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2898 DAHDI/ISDN supports call completion for the following switch types:
2899 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2900 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2902 Multicast RTP Support
2903 ---------------------
2904 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2905 The channel driver can be used with the Page application to perform multicast RTP
2906 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2907 Type can be either basic or linksys.
2908 Destination is the IP address and port for the RTP packets.
2909 Control address is specific to the linksys type and is used for sending the control
2910 packets unique to them.
2912 Security Events Framework
2913 -------------------------
2914 * Asterisk has a new C API for reporting security events. The module res_security_log
2915 sends these events to the "security" logger level. Currently, AMI is the only
2916 Asterisk component that reports security events. However, SIP support will be
2917 coming soon. For more information on the security events framework, see the
2918 "Asterisk Security Framework" section of the Asterisk wiki at
2919 https://wiki.asterisk.org/wiki/x/wgBQ
2920 * SIP support was added in Asterisk 10
2921 * This API now supports IPv6 addresses
2925 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2926 * A spandsp based fax backend (res_fax_spandsp) has been added.
2927 * The app_fax module has been deprecated in favor of the res_fax module and
2928 the new res_fax_spandsp backend.
2929 * The SendFAX and ReceiveFAX applications now send their log messages to a
2930 'fax' logger level, instead of to the generic logger levels. To see these
2931 messages, the system's logger.conf file will need to direct the 'fax' logger
2932 level to one or more destinations; the logger.conf.sample file includes an
2933 example of how to do this. Note that if the 'fax' logger level is *not*
2934 directed to at least one destination, log messages generated by these
2935 applications will be lost, and that if the 'fax' logger level is directed to
2936 the console, the 'core set verbose' and 'core set debug' CLI commands will
2937 have no effect on whether the messages appear on the console or not.
2941 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2942 Now, in order to enable transmitting silence during record the transmit_silence
2943 option should be used. transmit_silence_during_record remains a valid option, but
2944 defaults to the behavior of the transmit_silence option.
2945 * Addition of the Unit Test Framework API for managing registration and execution
2946 of unit tests with the purpose of verifying the operation of C functions.
2947 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2948 XMPP text messages to the remote JID.
2949 * Modules.conf has a new option - "require" - that marks a module as critical for
2950 the execution of Asterisk.
2951 If one of the required modules fail to load, Asterisk will exit with a return
2953 * An 'X' option has been added to the asterisk application which enables #exec support.
2954 This allows #exec to be used in asterisk.conf.
2955 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2956 * A new lockconfdir option has been added to asterisk.conf to protect the
2957 configuration directory (/etc/asterisk by default) during reloads.
2958 * The parkeddynamic option has been added to features.conf to enable the creation
2959 of dynamic parkinglots.
2960 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2961 the reportalarms config option.
2962 * chan_dahdi supports dialing configuring and dialing by device file name.
2963 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2964 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2965 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2966 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2967 Handy for the above name-based syntax as it does not depend on
2968 initialization order.
2969 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2970 significant increase in performance (about 3X) for installations using this switchtype.
2971 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2972 AIS. For more information, please see the Distributed Device State section of the
2973 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2974 * The addition of G.719 pass-through support.
2975 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2976 during device configuration.
2977 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2978 have less than 3 lines on the LCD.
2979 * Realtime now supports database failover. See the sample extconfig.conf for details.
2980 * The addition of improved translation path building for wideband codecs. Sample
2981 rate changes during translation are now avoided unless absolutely necessary.
2982 * The addition of the res_stun_monitor module for monitoring and reacting to network
2983 changes while behind a NAT.
2984 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2985 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2986 These allow support for any Administration. Default is AT&T values.
2990 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2991 optionally accept a filename, to apply the setting only to the code generated from
2992 that source file when Asterisk was built. However, there are some modules in Asterisk
2993 that are composed of multiple source files, so this did not result in the behavior
2994 that users expected. In this version, 'core set debug' and 'core set verbose'
2995 can optionally accept *module* names instead (with or without the .so extension),
2996 which applies the setting to the entire module specified, regardless of which source
2997 files it was built from.
2998 * New 'manager show settings' command showing the current settings loaded from
3000 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3001 the channel hangup request to all channels.
3002 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3004 ------------------------------------------------------------------------------
3005 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3006 ------------------------------------------------------------------------------
3010 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3011 Snom phones use this for call pickup of extensions that the phone is
3013 * Added support for setting the domain in the URI for caller of an
3014 outbound call by using the SIPFROMDOMAIN channel variable.
3015 * Added a new configuration option "remotesecret" for authentication to
3016 remote services. For backwards compatibility, "secret" still has the
3017 same function as before, but now you can configure both a remote secret and a
3018 local secret for mutual authentication.
3019 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3020 the sound will be played to the target of an attended transfer
3021 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3022 finer control over how many peers Asterisk will qualify and the gap between them
3023 when all peers need to be qualified at the same time.
3024 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3025 (either globally or for a specific peer), chan_sip will treat any SDP data
3026 it receives as new data and update the media stream accordingly. By
3027 default, Asterisk will only modify the media stream if the SDP session
3028 version received is different from the current SDP session version. This
3029 option is required to interoperate with devices that have non-standard SDP
3030 session version implementations (observed with Microsoft OCS). This option
3031 is disabled by default.
3032 * The parsing of register => lines in sip.conf has been modified to allow a port
3033 to be present in the "user" portion. Please see the sip.conf.sample file for more
3035 * Added support for subscribing to MWI on a remote server and making the status available
3036 as a mailbox. Please see the sip.conf.sample file for more information.
3037 * Added a function to remove SIP headers added in the dialplan before the
3038 first INVITE is generated - SIPRemoveHeader()
3039 * Channel variables set with setvar= in a device configuration is now
3040 set both for inbound and outbound calls.
3041 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3045 * Added immediate option to iax.conf
3046 * Added forceencryption option to iax.conf
3047 * Added Encryption and Trunk status to manager command "iaxpeers"
3051 * The configuration file now holds separate sections for devices and lines.
3052 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3057 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3058 support for LibOpenR2. http://www.libopenr2.org/
3059 * The UK option waitfordialtone has been added for use with BT analog
3061 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3062 is used in conjunction with the 'faxdetect' configuration option. When
3063 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3064 switch to the configured faxbuffers policy. For example, to use 6 buffers
3065 and a 'full' buffer policy for a fax transmission, add:
3067 The faxbuffers configuration will be in affect until the call is torn down.
3068 * Added service message support for 4ESS/5ESS switches.
3072 * For DAHDI channels, the CHANNEL() dialplan function now
3073 supports changing the channel's buffer policy (for the current
3074 call only), using this syntax:
3076 exten => s,n,Set(CHANNEL(buffers)=6,full)
3078 This would change the channel to the 'full' buffer policy and
3079 6 (six) buffers. Possible options for this setting are the same
3080 as those in chan_dahdi.conf.
3081 * Added a new dialplan function, CURLOPT, which permits setting various
3082 options that may be useful with the CURL dialplan function, such as
3083 cookies, proxies, connection timeouts, passwords, etc.
3084 * Permit the syntax and synopsis fields of the corresponding dialplan
3085 functions to be individually set from func_odbc.conf.
3086 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3087 * func_odbc now may specify an insert query to execute, when the write query
3088 affects 0 rows (usually indicating that no such row exists).
3089 * Added a new dialplan function, LISTFILTER, which permits removing elements
3090 from a set list, by name. Uses the same general syntax as the existing CUT
3091 and FIELDQTY dialplan functions, which also manage lists.
3092 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3093 obtaining realtime data from the dialplan.
3094 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3095 a subroutine when using the GoSub() and Return() applications.
3096 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3097 of "core show function AUDIOHOOK_INHERIT" from the CLI
3098 * Added AES_ENCRYPT. For information on its use, please see the output
3099 of "core show function AES_ENCRYPT" from the CLI
3100 * Added AES_DECRYPT. For information on its use, please see the output
3101 of "core show function AES_DECRYPT" from the CLI
3102 * func_odbc now supports database transactions across multiple queries.
3106 * Scheduled meetme conferences may now have their end times extended by
3108 * app_authenticate now gives the ability to select a prompt other than
3110 * app_directory now pays attention to the searchcontexts setting in
3111 voicemail.conf and will look through all contexts, if no context is
3112 specified in the initial argument.
3113 * A new application, Originate, has been introduced, that allows asynchronous
3114 call origination from the dialplan.
3115 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3116 in addition to the setting in the "general" context.
3117 * Added ConfBridge dialplan application which does conference bridges without
3118 DAHDI. For information on its use, please see the output of
3119 "core show application ConfBridge" from the CLI.
3123 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3124 operation to the AMI Redirect action.
3125 * extensions.conf now allows you to use keyword "same" to define an extension
3126 without actually specifying an extension. It uses exactly the same pattern
3127 as previously used on the last "exten" line. For example:
3128 exten => 123,1,NoOp(something)
3129 same => n,SomethingElse()
3130 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3131 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3132 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3133 by the new clialiases module. See cli_aliases.conf.sample file.
3134 * Times within timespecs are now accurate down to the minute. This is a change
3135 from historical Asterisk, which only provided timespecs rounded to the nearest
3136 even (read: evenly divisible by 2) minute mark.
3137 * The realtime switch now supports an option flag, 'p', which disables searches for
3139 * In addition to a time range and date range, timespecs now accept a 5th optional
3140 argument, timezone. This allows you to perform time checks on alternate
3141 timezones, especially if those daylight savings time ranges vary from your
3142 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3144 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3145 give you the correct output for an asterisk box behind nat. It will give you the
3146 externhost and localnet settings.
3147 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3148 can connect calls in passthrough mode, as well as record and play back files.
3149 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3150 using pickupsound and pickupfailsound in features.conf.
3151 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3152 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3153 instead of the /var/run/asterisk.pid where it used to be. This will make
3154 installs as non-root easier to manage.
3159 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3160 be written; they will no longer be explicitly written.
3162 Asterisk Manager Interface
3163 --------------------------
3164 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3165 a non-empty value) in your request. If you do this, any pending AMI events will
3166 *not* be included in the response to your request as they would normally, but
3167 will be left in the event queue for the next request you make to retrieve. For
3168 some applications, this will allow you to guarantee that you will only see
3169 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3170 To know whether the Asterisk server supports this header or not, your client can
3171 inspect the first response back from the server to see if it includes this header:
3173 Pragma: SuppressEvents
3175 If this is included, the server supports event suppression.
3177 * Added 4 new Actions to list skinny device(s) and line(s)
3183 LDAP Schema File Additions
3184 --------------------------
3185 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3186 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3188 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3189 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3190 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3191 * Removed redundant IPaddr (there's already IPAddress)
3192 - Gives more configuration Flags for SIP-Users available (tested)
3193 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3194 without extensibleObject (which really should be the last resort); gives
3195 also additional possibilities for LDAP-filter
3197 ------------------------------------------------------------------------------
3198 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3199 ------------------------------------------------------------------------------
3201 Device State Handling
3202 ---------------------
3203 * The event infrastructure in Asterisk got another big update to help support
3204 distributed events. It currently supports distributed device state and
3205 distributed Voicemail MWI (Message Waiting Indication). A new module has
3206 been merged, res_ais, which facilitates communicating events between servers.
3207 It uses the SAForum AIS (Service Availability Forum Application Interface
3208 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3209 a cluster of Asterisk servers, and to share events between them. For more
3210 information on setting this up, refer to the Distributed Device State section
3211 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3215 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3216 variables from an Asterisk configuration file.
3217 * The JACK_HOOK function now has a c() option to supply a custom client name.
3218 * Added two new dialplan functions from libspeex for audio gain control and
3219 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3220 rx directions of a channel from the dialplan.
3221 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3222 based on other parameters. The default is still to search based on the
3223 forwarding station ID. However, there are new options that allow you to search
3224 based on the message desk terminal ID, or the message desk number.
3225 * TIMEOUT() has been modified to be accurate down to the millisecond.
3226 * ENUM*() functions now include the following new options:
3227 - 'u' returns the full URI and does not strip off the URI-scheme.
3228 - 's' triggers ISN specific rewriting
3229 - 'i' looks for branches into an Infrastructure ENUM tree
3230 - 'd' for a direct DNS lookup without any flipping of digits.
3231 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3232 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3233 deviation of jitter, rtt, and loss for a call using chan_sip.
3235 DAHDI channel driver (chan_dahdi) Changes
3236 ----------------------------------------
3237 * Channels can now be configured using named sections in chan_dahdi.conf, just
3238 like other channel drivers, including the use of templates.
3239 * The default for pridialplan has changed from 'national' to 'unknown'.
3243 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3244 to something that matches the pattern a hint will be created using the contents
3245 and variables evaluated.
3246 * Dialplan matching has been extended to allow an extension to return to the
3247 PBX core to wait for more digits. This is done by using the new dialplan
3248 application called "Incomplete". This will permit a whole new level of
3249 extension control, by giving the administrator more control over early
3250 matches employing one of the short-circuit pattern match operators. Note
3251 that custom applications can trigger this same behavior by returning the
3252 special value AST_PBX_INCOMPLETE.
3256 * Directory now permits both first and last names to be matched at the same
3257 time. In addition, the number of digits to enter of the name can be set in
3258 the arguments to Directory; previously, you could enter only 3, regardless
3259 of how many names are in your company. For large companies, this should be
3261 * Voicemail now permits a mailbox setting to wrap around from first to last
3262 messages, if the "messagewrap" option is set to a true value.
3263 * Voicemail now permits an external script to be run, for password validation.
3264 The script should output "VALID" or "INVALID" on stdout, depending upon the
3265 wish to validate or invalidate the password given. Arguments are:
3266 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3268 * Dial has a new option: F(context^extension^pri), which permits a callee to
3269 continue in the dialplan, at the specified label, if the caller hangs up.
3270 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3271 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3272 * The Jack application now has a c() option to supply a custom client name.
3273 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3274 like the pre-existing whisper mode, except that the spy can also talk to the
3275 participant on the bridged channel as well.
3276 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3277 to be spoken instead of the channel name or number. For more information on the
3278 use of this option, issue the command "core show application ChanSpy" from the
3280 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3281 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3282 words, if using the 'd' option, it is not possible to enter a number to append to
3283 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3284 change to whisper mode, and pressing 6 will change to barge mode.
3285 * ExternalIVR now takes several options that affect the way it performs, as
3286 well as having several new commands. Please see the External IVR page on the Asterisk
3287 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3288 * Added ability to communicate over a TCP socket instead of forking a child process for the
3289 ExternalIVR application.
3290 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3291 of just the first one if you give the function more then one channel to check.
3292 * PrivacyManager now takes an option where you can specify a context where the
3293 given number will be matched. This way you have more control over who is allowed
3294 and it stops the people who blindly enter 10 digits.
3295 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3296 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3297 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3298 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3299 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3300 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3301 * The Dial() application no longer copies the language used by the caller to the callee's
3302 channel. If you desire for the caller's channel's language to be used for file playback
3303 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3304 * SendImage() no longer hangs up the channel on error; instead, it sets the
3305 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3306 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3308 * Park has a new option, 's', which silences the announcement of the parking space number.
3309 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3310 invalid input and will be assumed to mean that no timeout is desired.
3314 * Added DNS manager support to registrations for peers referencing peer entries.
3315 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3316 as well as periodically updating the IP address. These properties allow for
3317 better performance as well as recovery in the event of an IP change.
3318 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3319 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3320 These changes also provide performance improvements for call setup and tear down.
3321 * Added ability to specify registration expiry time on a per registration basis in
3323 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3325 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3326 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3327 * 'sip show peers' and 'sip show users' display their entries sorted in
3328 alphabetical order, as opposed to the order they were in, in the config
3330 * Videosupport now supports an additional option, "always", which always sets
3331 up video RTP ports, even on clients that don't support it. This helps with
3332 callfiles and certain transfers to ensure that if two video phones are
3333 connected, they will always share video feeds.
3337 * Existing DNS manager lookups extended to check for SRV records.
3338 * IAX2 encryption support has been improved to support periodic key rotation
3339 within a call for enhanced security. The option "keyrotate" has been
3340 provided to disable this functionality to preserve backwards compatibility
3341 with older versions of IAX2 that do not support key rotation.
3345 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3346 data tree based on the given <path>.
3347 * New CLI command "data show providers" that will display all the registered
3349 * New CLI command, "config reload <file.conf>" which reloads any module that
3350 references that particular configuration file. Also added "config list"
3351 which shows which configuration files are in use.
3352 * New CLI commands, "pri show version" and "ss7 show version" that will
3353 display which version of libpri and libss7 are being used, respectively.
3354 A new API call was added so trunk will now have to be compiled against
3355 a versions of libpri and libss7 that have them or it will not know that
3356 these libraries exist.
3357 * The commands "core show globals", "core set global" and "core set chanvar" has
3358 been deprecated in favor of the more semanticly correct "dialplan show globals",
3359 "dialplan set chanvar" and "dialplan set global".
3360 * New CLI command "dialplan show chanvar" to list all variables associated
3361 with a given channel.
3365 * Addresses managed by DNS manager now can check to see if there is a DNS
3366 SRV record for a given domain and will use that hostname/port if present.
3368 AMI - The manager (TCP/TLS/HTTP)
3369 --------------------------------
3370 * The Status command now takes an optional list of variables to display
3371 along with channel status.
3372 * The QueueEntry event now also includes the channel's uniqueid
3376 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3377 as some people were running into this limit. This limit has been increased
3382 * The TRANSFER queue log entry now includes the the caller's original
3383 position in the transferred-from queue.
3384 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3385 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3386 as well as an explanation about timeout options in general
3387 * Added a new option - C - for forcing the "answered elsewhere" flag on
3388 cancellation of calls in to members of the queue. This is to avoid the
3389 call to a member of a queue having the call listed as a "missed call".
3393 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3394 adaptive capabilities. What this means in practical terms is that if your
3395 realtime table lacks critical fields, Asterisk will now emit warnings to
3396 that effect. Also, some of the realtime drivers have the ability (if
3397 configured) to automatically add those columns to the table with the
3398 correct type and length.
3402 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3403 the 'setvar' option to cause a given audio file to be played upon completion
3404 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3405 Skinny channels only.
3406 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3407 Hoard page on the Asterisk wiki for more information:
3408 https://wiki.asterisk.org/wiki/x/pQBB
3409 * Config file variables may now be appended to, by using the '+=' append
3410 operator. This is most helpful when working with long SQL queries in
3411 func_odbc.conf, as the queries no longer need to be specified on a single
3413 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3414 which will add a second to the billsec when the ending
3415 time is set, if the number in the microseconds field of the end time is
3416 greater than the number of microseconds in the answer time. This allows
3417 users to count the 'initiated' seconds in their billing records.
3419 ------------------------------------------------------------------------------
3420 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3421 ------------------------------------------------------------------------------
3423 AMI - The manager (TCP/TLS/HTTP)
3424 --------------------------------
3425 * Manager has undergone a lot of changes, all of them documented
3426 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3427 * Manager version has changed to 1.1
3428 * Added a new action 'CoreShowChannels' to list currently defined channels
3429 and some information about them.
3430 * Added a new action 'SIPshowregistry' to list SIP registrations.
3431 * Added TLS support for the manager interface and HTTP server
3432 * Added the URI redirect option for the built-in HTTP server
3433 * The output of CallerID in Manager events is now more consistent.
3434 CallerIDNum is used for number and CallerIDName for name.
3435 * Enable https support for builtin web server.
3436 See configs/http.conf.sample for details.
3437 * Added a new action, GetConfigJSON, which can return the contents of an
3438 Asterisk configuration file in JSON format. This is intended to help
3439 improve the performance of AJAX applications using the manager interface
3441 * SIP and IAX manager events now use "ChannelType" in all cases where we
3442 indicate channel driver. Previously, we used a mixture of "Channel"
3443 and "ChannelDriver" headers.
3444 * Added a "Bridge" action which allows you to bridge any two channels that
3445 are currently active on the system.
3446 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3447 the voicemail users setup.
3448 * Added 'DBDel' and 'DBDelTree' manager commands.
3449 * cdr_manager now reports events via the "cdr" level, separating it from
3450 the very verbose "call" level.
3451 * Manager users are now stored in memory. If you change the manager account
3452 list (delete or add accounts) you need to reload manager.
3453 * Added Masquerade manager event for when a masquerade happens between
3455 * Added "manager reload" command for the CLI
3456 * Lots of commands that only provided information are now allowed under the
3457 Reporting privilege, instead of only under Call or System.
3458 * The IAX* commands now require either System or Reporting privilege, to
3459 mirror the privileges of the SIP* commands.
3460 * Added ability to retrieve list of categories in a config file.
3461 * Added ability to retrieve the content of a particular category.
3462 * Added ability to empty a context.
3463 * Created new action to create a new file.
3464 * Updated delete action to allow deletion by line number with respect to category.
3465 * Added new action insert to add new variable to category at specified line.
3466 * Updated action newcat to allow new category to be inserted in file above another
3468 * Added new event "JitterBufStats" in the IAX2 channel
3469 * Originate now requires the Originate privilege and, if you want to call out
3470 to a subshell, it requires the System privilege, as well. This was done to
3471 enhance manager security.
projects / asterisk/asterisk.git / blob