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 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
23 as the chanprefix parameter if the 'u' option is specified.
26 --------------------------
27 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
28 conference user menus.
30 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
31 menus, bridge settings, and user settings that have been applied by the
32 CONFBRIDGE dialplan function.
34 * The ConfBridge dialplan application now sets a channel variable,
35 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
36 how a channel exited the conference.
38 * Added conference user option 'announce_join_leave_review'. This option
39 implies 'announce_join_leave' with the added effect that the user will
40 be asked if they want to confirm or re-record the recording of their
41 name when entering the conference
44 --------------------------
45 * At exit, the Directory application now sets a channel variable
46 DIRECTORY_RESULT to one of the following based on the reason for exiting:
47 OPERATOR user requested operator by pressing '0' for operator
48 ASSISTANT user requested assistant by pressing '*' for assistant
49 TIMEOUT user pressed nothing and Directory stopped waiting
50 HANGUP user's channel hung up
51 SELECTED user selected a user from the directory and is routed
52 USEREXIT user pressed '#' from the selection prompt to exit
53 FAILED directory failed in a way that wasn't accounted for. Dang.
56 --------------------------
57 * MusicOnHold streams (all modes other than "files") now support wide band
61 --------------------------
62 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
63 and for the channel executing Page respectively.
66 --------------------------
67 * PickupChan now accepts channel uniqueids of channels to pickup.
70 --------------------------
71 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
72 to 'true' (case insensitive), then any Say application (SayNumber,
73 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
74 anticipate DTMF. If DTMF is received, these applications will behave like
75 the background application and jump to the received extension once a match
76 is established or after a short period of inactivity.
79 -------------------------
80 * A new function, MIXMONITOR, has been added to allow access to individual
81 instances of MixMonitor on a channel.
84 -------------------------
85 * Core Show Locks output now includes Thread/LWP ID if the platform
86 supports this feature.
87 * New "logger add channel" and "logger remove channel" CLI commands have
88 been added to allow creation and deletion of dynamic logger channels
89 without configuration changes. These dynamic logger channels will only
90 exist until the next restart of asterisk.
94 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
95 the new AST_SORCERY diaplan function.
99 * The live recording object on recording events now contains a target_uri
100 field which contains the URI of what is being recorded.
102 * The bridge type used when creating a bridge is now a comma separated list of
103 bridge properties. Valid options are: mixing, holding, dtmf_events, and
106 * A channelId can now be provided when creating a channel, either in the
107 uri (POST channels/my-channel-id) or as query parameter. A local channel
108 will suffix the second channel id with ';2' unless provided as query
109 parameter otherChannelId.
111 * A bridgeId can now be provided when creating a bridge, either in the uri
112 (POST bridges/my-bridge-id) or as a query parameter.
114 * A playbackId can be provided when starting a playback, either in the uri
115 (POST channels/my-channel-id/play/my-playback-id) or as a query parameter.
117 * A snoop channel can be started with a snoopId, in the uri or query.
121 * Originate now takes optional parameters ChannelId and OtherChannelId,
122 used to set the UniqueId on creation. The other id is assigned to the
123 second channel when dialing LOCAL, or defaults to appending ;2 if only
124 the single Id is given.
128 * A new set of Alembic scripts has been added for CDR tables. This will create
129 a 'cdr' table with the default schema that Asterisk expects.
133 * A new module, res_hep, has been added, that acts as a generic packet
134 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
135 It can be configured via hep.conf. Other modules can use res_hep to send
136 message traffic to a HEP capture server.
140 * A new module, res_hep_pjsip, has been added that will forward PJSIP
141 message traffic to a HEP capture server. See res_hep for more
146 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
147 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
149 * Added the following new CLI commands:
150 - "pjsip show contacts" - list all current PJSIP contacts.
151 - "pjsip show contact" - show specific information about a current PJSIP
153 - "pjsip show channel" - show detailed information about a PJSIP channel.
157 * A new module, res_pjsip_multihomed handles situations where the system
158 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
159 determines which interface should be used during message sending.
161 res_pjsip_pidf_digium_body_supplement
163 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
164 request body formatting for presence support in Digium phones.
166 res_pjsip_send_to_voicemail
168 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
169 particular headers to transfer a PJSIP channel directly to a particular
170 extension that has VoiceMail. This is intended to be used with Digium
171 phones that support this feature.
173 res_pjsip_outbound_registration
175 * A new CLI command has been added: "pjsip show registrations", which lists
176 all configured PJSIP registrations
179 ------------------------------------------------------------------------------
180 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
181 ------------------------------------------------------------------------------
185 * Added a new module that provides AMI control over MWI within Asterisk,
186 res_mwi_external_ami. Note that this module depends on res_mwi_external;
187 for more information on enabling this module, see res_mwi_external.
188 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
189 the MWIGet/MWIGetComplete events.
191 * The DialStatus field in the DialEnd event can now contain additional
192 statuses that convey how the dial operation terminated. This includes
193 ABORT, CONTINUE, and GOTO.
195 * AMI will now emit security events. A new class authorization has been
196 added in manager.conf for the security events, 'security'. The new events
198 - FailedACL - raised when a request violates an ACL check
199 - InvalidAccountID - raised when a request fails an authentication
200 check due to an invalid account ID
201 - SessionLimit - raised when a request fails due to exceeding the
202 number of allowed concurrent sessions for a service
203 - MemoryLimit - raised when a request fails due to an internal memory
205 - LoadAverageLimit - raised when a request fails because a configured
206 load average limit has been reached
207 - RequestNotAllowed - raised when a request is not allowed by
209 - AuthMethodNotAllowed - raised when a request used an authentication
210 method not allowed by the service
211 - RequestBadFormat - raised when a request is received with bad formatting
212 - SuccessfulAuth - raised when a request successfully authenticates
213 - UnexpectedAddress - raised when a request has a different source address
214 then what is expected for a session already in progress with a service
215 - ChallengeResponseFailed - raised when a request's attempt to authenticate
216 has been challenged, and the request failed the authentication challenge
217 - InvalidPassword - raised when a request provides an invalid password
218 during an authentication attempt
219 - ChallengeSent - raised when an Asterisk service send an authentication
220 challenge to a request
221 - InvalidTransport - raised when a request attempts to use a transport not
222 allowed by the Asterisk service
224 * Bridge related events now have two additional fields: BridgeName and
225 BridgeCreator. BridgeName is a descriptive name for the bridge;
226 BridgeCreator is the name of the entity that created the bridge. This
227 affects the following events: ConfbridgeStart, ConfbridgeEnd,
228 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
229 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
230 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
234 * The Bridge data model now contains the additional fields 'name' and
235 'creator'. The 'name' field conveys a descriptive name for the bridge;
236 the 'creator' field conveys the name of the entity that created the bridge.
237 This affects all responses to HTTP requests that return a Bridge data model
238 as well as all event derived data models that contain a Bridge data model.
239 The POST /bridges operation may now optionally specify a name to give to
240 the bridge being created.
242 * Added a new ARI resource 'mailboxes' which allows the creation and
243 modification of mailboxes managed by external MWI. Modules res_mwi_external
244 and res_stasis_mailbox must be enabled to use this resource. For more
245 information on external MWI control, see res_mwi_external.
247 * Added new events for externally initiated transfers. The event
248 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
249 of a bridge in the ARI controlled application to the dialplan; the
250 BridgeAttendedTransfer event is raised when a channel initiates an
251 attended transfer of a bridge in the ARI controlled application to the
254 * Channel variables may now be specified as a body parameter to the
255 POST /channels operation. The 'variables' key in the JSON is interpreted
256 as a sequence of key/value pairs that will be added to the created channel
257 as channel variables. Other parameters in the JSON body are treated as
258 query parameters of the same name.
262 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
263 automatically handled by the HTTP server if a request is received with a
264 Transfer-Encoding type of "chunked".
268 * Path support has been added with the 'support_path' option in registration
271 * A 'debug' option has been added to the globals section that will allow
272 sip messages to be logged.
274 * A 'set_var' option has been added to endpoints that will automatically
275 set the desired variable(s) on a channel created for that endpoint.
277 * Several new tables and columns have been added to the realtime schema for
278 the res_pjsip related modules. See the UPGRADE.txt notes for updating
283 * A new module, res_mwi_external, has been added to Asterisk. This module
284 acts as a base framework that other modules can build on top of to allow
285 an external system to control MWI within Asterisk. For implementations
286 that make use of res_mwi_external, see res_mwi_external_ami and
287 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
288 that may produce MWI themselves, such as app_voicemail. res_mwi_external
289 and other modules that depend on it cannot be built or loaded with
290 app_voicemail present.
294 * DNS functionality will now automatically be enabled if the system configured
295 nameservers can be retrieved. If the system configured nameservers can not be
296 retrieved the functionality will resort to using system resolution. Functionalty
297 such as SRV records and failover will not be available if system resolution
300 ------------------------------------------------------------------------------
301 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
302 ------------------------------------------------------------------------------
307 Asterisk 12 is a standard release of the Asterisk project. As such, the
308 focus of development for this release was on core architectural changes and
309 major new features. This includes:
310 * A more flexible bridging core based on the Bridging API
311 * A new internal message bus, Stasis
312 * Major standardization and consistency improvements to AMI
313 * Addition of the Asterisk RESTful Interface (ARI)
314 * A new SIP channel driver, chan_pjsip
315 In addition, as the vast majority of bridging in Asterisk was migrated to the
316 Bridging API used by ConfBridge, major changes were made to most of the
317 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
319 Specifications have been written for the affected interfaces. These
320 specifications are available on the Asterisk wiki:
321 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
322 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
323 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
325 It is *highly* recommended that anyone migrating to Asterisk 12 read the
326 information regarding its release both in this file and in the accompanying
327 UPGRADE.txt file. More detailed information on the major changes can be found
328 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
333 * Added build option DISABLE_INLINE. This option can be used to work around a
334 bug in gcc. For more information, see
335 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
337 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
338 the CHANNEL_TRACE build option were incompatible with the new bridging
341 * Asterisk now optionally uses libxslt to improve XML documentation generation
342 and maintainability. If libxslt is not available on the system, some XML
343 documentation will be incomplete.
345 * Asterisk now depends on libjansson. If a package of libjansson is not
346 available on your distro, please see http://www.digip.org/jansson/.
348 * Asterisk now depends on libuuid and, optionally, uriparser. It is
349 recommended that you install uriparser, even if it is optional.
351 * The new SIP stack and channel driver uses a particular version of PJSIP.
352 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
353 configuring and installing PJSIP for usage with Asterisk.
355 * Optional API was re-implemented to be more portable, and no longer requires
356 weak reference support from the compiler. The build option OPTIONAL_API may
357 be disabled to disable Optional API support.
364 * Along with AgentRequest, this application has been modified to be a
365 replacement for chan_agent. The act of a channel calling the AgentLogin
366 application places the channel into a pool of agents that can be
367 requested by the AgentRequest application. Note that this application, as
368 well as all other agent related functionality, is now provided by the
369 app_agent_pool module. See chan_agent and AgentRequest for more information.
371 * This application no longer performs agent authentication. If authentication
372 is desired, the dialplan needs to perform this function using the
373 Authenticate or VMAuthenticate application or through an AGI script before
376 * If this application is called and the agent is already logged in, the
377 dialplan will continue exection with the AGENT_STATUS channel variable set
378 to ALREADY_LOGGED_IN.
380 * The agents.conf schema has changed. Rather than specifying agents on a
381 single line in comma delineated fashion, each agent is defined in a separate
382 context. This allows agents to use the power of context templates in their
385 * A number of parameters from agents.conf have been removed. This includes
386 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
387 urlprefix, and savecallsin. These options were obsoleted by the move from
388 a channel driver model to the bridging/application model provided by
393 * A new application, this will request a logged in agent from the pool and
394 bridge the requested channel with the channel calling this application.
395 Logged in agents are those channels that called the AgentLogin application.
396 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
397 application will be set with an appropriate error value.
401 * This application has been removed. It was a holdover from when
402 AgentCallbackLogin was removed.
406 * Added support for additional Ademco DTMF signalling formats, including
407 Express 4+1, Express 4+2, High Speed and Super Fast.
409 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
410 call time, in milliseconds, to run the application.
412 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
413 maximum number of times to retry the call.
415 * Added a new configuration option answait. If set, the AlarmReceiver
416 application will wait the number of milliseconds specified by answait
417 after the channel has answered. Valid values range between 500
418 milliseconds and 10000 milliseconds.
420 * Added configuration option no_group_meta. If enabled, grouping of metadata
421 information in the AlarmReceiver log file will be skipped.
425 * It is now no longer possible to bypass updating the CDR on the channel
426 when answering. CDRs reflect the state of the channel and will always
427 reflect the time they were Answered.
431 * A new application in Asterisk, this will place the calling channel
432 into a holding bridge, optionally entertaining them with some form of
433 media. Channels participating in a holding bridge do not interact with
434 other channels in the same holding bridge. Optionally, however, a channel
435 may join as an announcer. Any media passed from an announcer channel is
436 played to all channels in the holding bridge. Channels leave a holding
437 bridge either when an optional timer expires, or via the ChannelRedirect
438 application or AMI Redirect action.
442 * All participants in a bridge can now be kicked out of a conference room
443 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
444 command, i.e., 'confbridge kick <conference> all'
446 * CLI output for the 'confbridge list' command has been improved. When
447 displaying information about a particular bridge, flags will now be shown
448 for the participating users indicating properties of that user.
450 * The ConfbridgeList event now contains the following fields: WaitMarked,
451 EndMarked, and Waiting. This displays additional properties about the
452 user's profile, as well as whether or not the user is waiting for a
453 Marked user to enter the conference.
455 * Added a new option for conference recording, record_file_append. If enabled,
456 when the recording is stopped and then re-started, the existing recording
457 will be used and appended to.
459 * ConfBridge now has the ability to set the language of announcements to the
460 conference. The language can be set on a bridge profile in confbridge.conf
461 or by the dialplan function CONFBRIDGE(bridge,language)=en.
465 * The channel variable CPLAYBACKSTATUS may now return the value
466 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
467 such as AMI. See the AMI action ControlPlayback for more information.
471 * Added the 'a' option, which allows the caller to enter in an additional
472 alias for the user in the directory. This option must be used in conjunction
473 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
474 specified in voicemail.conf.
478 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
479 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
480 containing the unique ID of the bridge that the channel happens to be in.
484 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
485 for more information.
487 * Variables are no longer purged from the original CDR. See the 'v' option for
490 * The 'A' option has been removed. The Answer time on a CDR is never updated
493 * The 'd' option has been removed. The disposition on a CDR is a function of
494 the state of the channel and cannot be altered.
496 * The 'D' option has been removed. Who the Party B is on a CDR is a function
497 of the state of the respective channels involved in the CDR and cannot be
500 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
501 such that the start time and, if applicable, the answer time was updated.
502 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
503 'r' option now triggers the Reset, setting the start time (and answer time
504 if applicable) to the current time. Note that the 'a' option still sets
505 the answer time to the current time if the channel was already answered.
507 * The 's' option has been removed. A variable can be set on the original CDR
508 if desired using the CDR function, and removed from a forked CDR using the
511 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
512 longer applies in the CDR engine.
514 * The 'v' option now prevents the copy of the variables from the original CDR
515 to the forked CDR. Previously the variables were always copied but were
516 removed from the original. This was changed as removing variables from a CDR
517 can have unintended side effects - this option allows the user to prevent
518 propagation of variables from the original to the forked without modifying
523 * Added the 'n' option to MeetMe to prevent application of the DENOISE
524 function to a channel joining a conference. Some channel drivers that vary
525 the number of audio samples in a voice frame will experience significant
526 quality problems if a denoiser is attached to the channel; this option gives
527 them the ability to remove the denoiser without having to unload func_speex.
531 * The 'b' option now includes conferences as well as sounds played to the
534 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
535 running during a transfer. If a MixMonitor is started on a channel,
536 the MixMonitor will continue to record the audio passing through the
537 channel even in the presence of transfers.
541 * The NoCDR application is deprecated. Please use the CDR_PROP function to
544 * While the NoCDR application will prevent CDRs for a channel from being
545 propagated to registered CDR backends, it will not prevent that data from
546 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
547 function that enables CDRs on a channel will restore those records that have
548 not yet been finalized.
552 * The app_parkandannounce module has been removed. The application
553 ParkAndAnnounce is now provided by the res_parking module. See the
554 res_parking changes for more information.
558 * Added queue available hint. The hint can be added to the dialplan using the
559 following syntax: exten,hint,Queue:{queue_name}_avail
560 For example, if the name of the queue is 'markq':
561 exten => 8501,hint,Queue:markq_avail
562 This will report 'InUse' if there are no logged in agents or no free agents.
563 It will report 'Idle' when an agent is free.
565 * Queues now support a hint for member paused state. The hint uses the form
566 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
567 are the name of the queue and the name of the member to subscribe to,
568 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
569 Members will show as In Use when paused.
571 * The configuration options eventwhencalled and eventmemberstatus have been
572 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
573 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
574 sent. The "Variable" fields will also no longer exist on the Agent* events.
575 These events can be filtered out from a connected AMI client using the
576 eventfilter setting in manager.conf.
578 * The queue log now differentiates between blind and attended transfers. A
579 blind transfer will result in a BLINDTRANSFER message with the destination
580 context and extension. An attended transfer will result in an
581 ATTENDEDTRANSFER message. This message will indicate the method by which
582 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
583 for running an application on a bridge or channel, or "LINK" for linking
584 two bridges together with local channels. The queue log will also now detect
585 externally initiated blind and attended transfers and record the transfer
588 * When performing queue pause/unpause on an interface without specifying an
589 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
590 least one member of any queue exists for that interface.
592 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
593 for realtime queue log entries.
597 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
598 CDRs when they were previously disabled on a channel.
600 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
601 backends occurs on an as-needed basis in order to preserve linkedid
602 propagation and other needed behavior.
606 * A new application, this is similar to SayAlpha except that it supports
607 case sensitive playback of the specified characters. For example,
608 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
612 * This application is deprecated in favor of CHANNEL(amaflags).
616 * The SendDTMF application will now accept 'W' as valid input. This will cause
617 the application to delay one second while streaming DTMF.
621 * A new application in Asterisk 12, this hands control of the channel calling
622 the application over to an external system. Currently, external systems
623 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
627 * UserEvent will now handle duplicate keys by overwriting the previous value
630 * In addition to AMI, UserEvent invocations will now be distributed to any
631 interested Stasis applications.
635 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
636 system as mailbox@context. The rest of the system cannot add @default
637 to mailbox identifiers for app_voicemail that do not specify a context
638 any longer. It is a mailbox identifier format that should only be
639 interpreted by app_voicemail.
641 * The voicemail.conf configuration file now has an 'alias' configuration
642 parameter for use with the Directory application. The voicemail realtime
643 database table schema has also been updated with an 'alias' column.
648 * Pass through support has been added for both VP8 and Opus.
650 * Added format attribute negotiation for the Opus codec. Format attribute
651 negotiation is provided by the res_format_attr_opus module.
656 * Masquerades as an operation inside Asterisk have been effectively hidden
657 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
658 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
659 dropping of frame/audio hooks, and other internal implementation details
660 that users had to deal with. This fundamental change has large implications
661 throughout the changes documented for this version. For more information
662 about the new core architecture of Asterisk, please see the Asterisk wiki.
664 * Multiple parties in a bridge may now be transferred. If a participant in a
665 multi-party bridge initiates a blind transfer, a Local channel will be used
666 to execute the dialplan location that the transferer sent the parties to. If
667 a participant in a multi-party bridge initiates an attended transfer,
668 several options are possible. If the attended transfer results in a transfer
669 to an application, a Local channel is used. If the attended transfer results
670 in a transfer to another channel, the resulting channels will be merged into
673 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
674 driver specific. If the channel variable is set on the transferrer channel,
675 the sound will be played to the target of an attended transfer.
677 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
678 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
679 listed. Any more peers in the bridge will not be included in the list.
680 BRIDGEPEER is not valid in holding bridges like parking since those channels
681 do not talk to each other even though they are in a bridge.
683 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
684 and will contain a value if the BRIDGEPEER's channel driver supports it.
686 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
687 was responsible for an attended transfer in a similar fashion to
690 * Modules using the Configuration Framework or Sorcery must have XML
691 configuration documentation. This configuration documentation is included
692 with the rest of Asterisk's XML documentation, and is accessible via CLI
693 commands. See the CLI changes for more information.
695 AMI (Asterisk Manager Interface)
697 * Major changes were made to both the syntax as well as the semantics of the
698 AMI protocol. In particular, AMI events have been substantially improved
699 in this version of Asterisk. For more information, please see the AMI
700 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
702 * AMI events that reference a particular channel or bridge will now always
703 contain a standard set of fields. When multiple channels or bridges are
704 referenced in an event, fields for at least some subset of the channels
705 and bridges in the event will be prefixed with a descriptive name to avoid
706 name collisions. See the AMI event documentation on the Asterisk wiki for
709 * The CLI command 'manager show commands' no longer truncates command names
710 longer than 15 characters and no longer shows authorization requirement
711 for commands. 'manager show command' now displays the privileges needed
712 for using a given manager command instead.
714 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
715 peer in its response if the peer has a subscribe context set.
717 * The SIPqualifypeer action now acknowledges the request once it has
718 established that the request is against a known peer. It also issues a new
719 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
721 * The PlayDTMF action now supports an optional 'Duration' parameter. This
722 specifies the duration of the digit to be played, in milliseconds.
724 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
725 updates when changes occur instead of requiring the use of pollmailboxes.
727 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
728 AMI client to manipulate audio currently being played back on a channel. The
729 supported operations depend on the application being used to send audio to
730 the channel. When the audio playback was initiated using the ControlPlayback
731 application or CONTROL STREAM FILE AGI command, the audio can be paused,
732 stopped, restarted, reversed, or skipped forward. When initiated by other
733 mechanisms (such as the Playback application), the audio can be stopped,
734 reversed, or skipped forward.
736 * Channel related events now contain a snapshot of channel state, adding new
737 fields to many of these events.
739 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
740 in a future release. Please use the common 'Exten' field instead.
742 * The AMI event 'UserEvent' from app_userevent now contains the channel state
743 fields. The channel state fields will come before the body fields.
745 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
746 'UnParkedCall' have changed significantly in the new res_parking module.
748 The 'Channel' and 'From' headers are gone. For the channel that was parked
749 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
750 has a number of fields associated with it. The old 'Channel' header relayed
751 the same data as the new 'ParkeeChannel' header.
753 The 'From' field was ambiguous and changed meaning depending on the event.
754 for most of these, it was the name of the channel that parked the call
755 (the 'Parker'). There is no longer a header that provides this channel name,
756 however the 'ParkerDialString' will contain a dialstring to redial the
757 device that parked the call.
759 On UnParkedCall events, the 'From' header would instead represent the
760 channel responsible for retrieving the parkee. It receives a channel
761 snapshot labeled 'Retriever'. The 'from' field is is replaced with
764 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
766 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
767 fashion has changed the field names 'StartExten' and 'StopExten' to
768 'StartSpace' and 'StopSpace' respectively.
770 * The deprecated use of | (pipe) as a separator in the channelvars setting in
771 manager.conf has been removed.
773 * Channel Variables conveyed with a channel no longer contain the name of the
774 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
775 ChanVariable: bar=baz. When multiple channels are present in a single AMI
776 event, the various ChanVariable fields will contain a suffix that specifies
777 which channel they correspond to.
779 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
780 event always conveys the AMI event for a particular channel.
782 * All 'Reload' events have been consolidated into a single event type. This
783 event will always contain a Module field specifying the name of the module
784 and a Status field denoting the result of the reload. All modules now issue
785 this event when being reloaded.
787 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
788 fail to receive this event due to being connected after modules have loaded.
789 AMI connections that want to know when Asterisk is ready should listen for
790 the 'FullyBooted' event.
792 * app_fax now sends the same send fax/receive fax events as res_fax. The
793 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
794 now the 'ReceiveFAX' event.
796 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
797 'MusicOnHoldStop'. The sub type field has been removed.
799 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
800 carrier for another protocol.
802 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
803 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
804 to the specific channel. 'Both' may be specified to play a tone to both
805 channels. The old 'yes' option is still accepted as a way of playing the
806 tone to Channel2 only.
808 * The AMI 'Status' response event to the AMI Status action replaces the
809 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
810 indicate what bridge the channel is currently in.
812 * The AMI 'Hold' event has been moved out of individual channel drivers, into
813 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
816 * The AMI events in app_queue have been made more consistent with each other.
817 Events that reference channels (QueueCaller* and Agent*) will show
818 information about each channel. The (infamous) 'Join' and 'Leave' AMI
819 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
821 * The 'MCID' AMI event now publishes a channel snapshot when available and
822 its non-channel-snapshot parameters now use either the "MCallerID" or
823 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
824 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
825 parameters in the channel snapshot.
827 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
828 'AgentLogin' and 'AgentLogoff' respectively.
830 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
831 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
833 * 'ChannelUpdate' events have been removed.
835 * All AMI events now contain a 'SystemName' field, if available.
837 * Local channel optimization is now conveyed in two events:
838 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
839 when the Local channel driver begins attempting to optimize itself out of
840 the media path; the End event is sent after the channel halves have
841 successfully optimized themselves out of the media path.
843 * Local channel information in events is now prefixed with 'LocalOne' and
844 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
845 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
846 and 'LocalOptimizationEnd' events.
848 * The option 'allowmultiplelogin' can now be set or overriden in a particular
849 account. When set in the general context, it will act as the default
850 setting for defined accounts.
852 * The 'BridgeAction' event was removed. It technically added no value, as the
853 Bridge Action already receives confirmation of the bridge through a
854 successful completion Event.
856 * The 'BridgeExec' events were removed. These events duplicated the events that
857 occur in the Briding API, and are conveyed now through BridgeCreate,
858 BridgeEnter, and BridgeLeave events.
860 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
861 previous versions. They now report all SR/RR packets sent/received, and
862 have been restructured to better reflect the data sent in a SR/RR. In
863 particular, the event structure now supports multiple report blocks.
865 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
866 raised when a blind transfer/attended transfer completes successfully.
867 They contain information about the transfer that just completed, including
868 the location of the transfered channel.
870 * Added a 'security' class to AMI which outputs the required fields for
871 security messages similar to the log messages from res_security_log
873 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
874 that describes the status value in a human readable string.
876 CDR (Call Detail Records)
878 * Significant changes have been made to the behavior of CDRs. The CDR engine
879 was effectively rewritten and built on the Stasis message bus. For a full
880 definition of CDR behavior in Asterisk 12, please read the specification
881 on the Asterisk wiki (wiki.asterisk.org).
883 * CDRs will now be created between all participants in a bridge. For each
884 pair of channels in a bridge, a CDR is created to represent the path of
885 communication between those two endpoints. This lets an end user choose who
886 to bill for what during bridge operations with multiple parties.
888 * The duration, billsec, start, answer, and end times now reflect the times
889 associated with the current CDR for the channel, as opposed to a cumulative
890 measurement of all CDRs for that channel.
892 * When a CDR is dispatched, user defined CDR variables from both parties are
893 included in the resulting CDR. If both parties have the same variable, only
894 the Party A value is provided.
896 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
897 information regarding the CDR engine is logged as verbose messages. This
898 option should only be used if the behavior of the CDR engine needs to be
901 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
902 normally configured in cdr.conf.
904 * Added CLI command 'cdr show active {channel}'. When {channel} is not
905 specified, this command provides a summary of the channels with CDR
906 information and their statistics. When {channel} is specified, it shows
907 detailed information about all records associated with {channel}.
909 CEL (Channel Event Logging)
911 * CEL has undergone significant rework in Asterisk 12, and is now built on the
912 Stasis message bus. Please see the specification for CEL on the Asterisk
913 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
916 * The 'extra' field of all CEL events that use it now consists of a JSON blob
917 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
919 * BLINDTRANSFER events now report the transferee bridge unique
920 identifier, extension, and context in a JSON blob as the extra string
921 instead of the transferee channel name as the peer.
923 * ATTENDEDTRANSFER events now report the peer as NULL and additional
924 information in the 'extra' string as a JSON blob. For transfers that occur
925 between two bridged channels, the 'extra' JSON blob contains the primary
926 bridge unique identifier, the secondary channel name, and the secondary
927 bridge unique identifier. For transfers that occur between a bridged channel
928 and a channel running an app, the 'extra' JSON blob contains the primary
929 bridge unique identifier, the secondary channel name, and the app name.
931 * LOCAL_OPTIMIZE events have been added to convey local channel
932 optimizations with the record occurring for the semi-one channel and
933 the semi-two channel name in the peer field.
935 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
936 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
937 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
938 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
939 regardless of whether or not that bridge happens to contain multiple
944 * When compiled with '--enable-dev-mode', the astobj2 library will now add
945 several CLI commands that allow for inspection of ao2 containers that
946 register themselves with astobj2. The CLI commands are 'astobj2 container
947 dump', 'astobj2 container stats', and 'astobj2 container check'.
949 * Added specific CLI commands for bridge inspection. This includes 'bridge
950 show all', which lists all bridges in the system, and 'bridge show {id}',
951 which provides specific information about a bridge.
953 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
954 ejecting the channels currently in the bridge. If the channels cannot
955 continue in the dialplan or application that put them in the bridge, they
958 * Added command 'bridge kick'. This will eject a single channel from a bridge.
960 * Added commands to inspect and manipulate the registered bridge technologies.
961 This include 'bridge technology show', which lists the registered bridge
962 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
963 which controls whether or not a registered bridge technology can be used
964 during smart bridge operations. If a technology is suspended, it will not
965 be used when a bridge technology is picked for channels; when unsuspended,
966 it can be used again.
968 * The command 'config show help {module} {type} {option}' will show
969 configuration documentation for modules with XML configuration
970 documentation. When {module}, {type}, and {option} are omitted, a listing
971 of all modules with registered documentation is displayed. When {module}
972 is specified, a listing of all configuration types for that module is
973 displayed, along with their synopsis. When {module} and {type} are
974 specified, a listing of all configuration options for that type are
975 displayed along with their synopsis. When {module}, {type}, and {option}
976 are specified, detailed information for that configuration option is
979 * Added 'core show sounds' and 'core show sound' CLI commands. These display
980 a listing of all installed media sounds available on the system and
981 detailed information about a sound, respectively.
983 * 'xmldoc dump' has been added. This CLI command will dump the XML
984 documentation DOM as a string to the specified file. The Asterisk core
985 will populate certain XML elements pulled from the source files with
986 additional run-time information; this command lets a user produce the
987 XML documentation with all information.
991 * Parking has been pulled from core and placed into a separate module called
992 res_parking. See Parking changes below for more details. Configuration for
993 parking should now be performed in res_parking.conf. Configuration for
994 parking in features.conf is now unsupported.
996 * Core attended transfers now have several new options. While performing an
997 attended transfer, the transferer now has the following options:
998 - *1 - cancel the attended transfer (configurable via atxferabort)
999 - *2 - complete the attended transfer, dropping out of the call
1000 (configurable via atxfercomplete)
1001 - *3 - complete the attended transfer, but stay in the call. This will turn
1002 the call into a multi-party bridge (configurable via atxferthreeway)
1003 - *4 - swap to the other party. Once an attended transfer has begun, this
1004 options may be used multiple times (configurable via atxferswap)
1006 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1007 must be on the channel initiating the transfer to have any effect.
1009 * The BRIDGE_FEATURES channel variable would previously only set features for
1010 the calling party and would set this feature regardless of whether the
1011 feature was in caps or in lowercase. Use of a caps feature for a letter
1012 will now apply the feature to the calling party while use of a lowercase
1013 letter will apply that feature to the called party.
1015 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1017 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1018 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1019 activated the dynamic feature.
1021 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1022 only on the channel executing the dynamic feature. Executing a dynamic
1023 feature on the bridge peer in a multi-party bridge will execute it on all
1024 peers of the activating channel.
1026 * You can now have the settings for a channel updated using the FEATURE()
1027 and FEATUREMAP() functions inherited to child channels by setting
1028 FEATURE(inherit)=yes.
1030 * automixmon now supports additional channel variables from automon including:
1031 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1032 and TOUCH_MIXMONITOR_MESSAGE_STOP
1034 * A new general features.conf option 'recordingfailsound' has been added which
1035 allowssetting a failure sound for a user tries to invoke a recording feature
1036 such as automon or automixmon and it fails.
1038 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1039 features.c for atxferdropcall=no to work properly. This option now just
1044 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1045 be used. Given that this can cause the Asterisk log files to grow quickly,
1046 this option should only be used if an external mechanism for log management
1051 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1052 will store the path information for that peer when it registers. Realtime
1053 tables can also use the 'supportpath' field to enable Path header support.
1055 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1056 objectIdentifier. This maps to the supportpath option in sip.conf.
1060 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1061 provides modules a useful abstraction on top of the many storage mechanisms
1062 in Asterisk, including the Asterisk Database, static configuration files,
1063 static Realtime, and dynamic Realtime. It also provides a caching service.
1064 Users can configure a hierarchy of data storage layers for specific modules
1067 * All future modules which utilize Sorcery for object persistence must have a
1068 column named "id" within their schema when using the Sorcery realtime module.
1069 This column must be able to contain a string of up to 128 characters in length.
1071 Security Events Framework
1073 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1074 the "seconds-microseconds" format that it was using previously.
1078 * The Stasis message bus is a publish/subscribe message bus internal to
1079 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1080 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1081 Stasis can be configured in stasis.conf. Note that these parameters operate
1082 at a very low level in Asterisk, and generally will not require changes.
1086 * When a channel driver is configured to enable jiterbuffers, they are now
1087 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1088 is already set for that channel when it enters, such as by the JITTERBUFFER
1089 function, then the existing jitterbuffer will be used and the one set by
1090 the channel driver will not be applied.
1094 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1095 dialplan applications provided by the app_agent_pool module. Agents are
1096 connected with callers using the new AgentRequest dialplan application.
1097 The Agents:<agent-id> device state is available to monitor the status of an
1098 agent. See agents.conf.sample for valid configuration options.
1100 * The updatecdr option has been removed. Altering the names of channels on a
1101 CDR is not supported - the name of the channel is the name of the channel,
1102 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1103 has also been removed, for the same reason.
1105 * The endcall and enddtmf configuration options are removed. Use the
1106 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1107 channel before calling AgentLogin.
1111 * chan_bridge has been removed. Its functionality has been incorporated
1112 directly into the ConfBridge application itself.
1116 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1117 of the specified span and its B-channels. Note that this command should
1118 only be used if you understand the risks it entails.
1120 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1121 A range of channels can be specified to be destroyed. Note that this command
1122 should only be used if you understand the risks it entails.
1124 * Added the CLI command 'dahdi create channels'. A range of channels can be
1125 specified to be created, or the keyword 'new' can be used to add channels
1128 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1129 the exact configured mailbox name. For app_voicemail mailboxes this is
1132 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1136 * IPv6 support has been added. We are now able to bind to and
1137 communicate using IPv6 addresses.
1141 * The /b option has been removed.
1143 * chan_local moved into the system core and is no longer a loadable module.
1147 * Added general support for busy detection.
1149 * Added ECAM command support for Sony Ericsson phones.
1153 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1154 SIP stack. A collection of resource modules provides the bulk of the SIP
1155 functionality. For more information on the new SIP channel driver, see
1156 https://wiki.asterisk.org/wiki/x/JYGLAQ
1160 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1161 using the 'supportpath' setting, either on a global basis or on a peer basis.
1162 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1163 set of proxies by using a pre-loaded route-set defined by the Path headers in
1164 the REGISTER request. See Realtime updates for more configuration information.
1166 * The SIP_CODEC family of variables may now specify more than one codec. Each
1167 codec must be separated by a comma. The first codec specified is the
1168 preferred codec for the offer. This allows a dialplan writer to specify both
1169 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1171 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1172 in the core, and can be filtered out using the 'eventfilter' parameter
1175 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1176 codecs configured for a peer instead of the requested codec.
1178 * The option "register_retry_403" has been added to chan_sip to work around
1179 servers that are known to erroneously send 403 in response to valid
1180 REGISTER requests and allows Asterisk to continue attepmting to connect.
1184 * Added the 'immeddialkey' parameter. If set, when the user presses the
1185 configured key the already entered number will be immediately dialed. This
1186 is useful when the dialplan allows for variable length pattern matching.
1187 Valid options are '*' and '#'.
1189 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1190 milliseconds) before a call forward is considered to not be answered.
1192 * The 'serviceurl' parameter allows Service URLs to be attached to line
1201 * The password option has been disabled, as the AgentLogin application no
1202 longer provides authentication.
1206 * Due to changes in the Asterisk core, this function is no longer needed to
1207 preserve a MixMonitor on a channel during transfer operations and dialplan
1208 execution. It is effectively obsolete.
1212 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1213 deprecated. Use the CHANNEL function instead to access these attributes.
1215 * The 'l' option has been removed. When reading a CDR attribute, the most
1216 recent record is always used. When writing a CDR attribute, all non-finalized
1219 * The 'r' option has been removed, for the same reason as the 'l' option.
1221 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1226 * A new function CDR_PROP has been added. This function lets you set properties
1227 on a channel's active CDRs. This function is write-only. Properties accept
1228 boolean values to set/clear them on the channel's CDRs. Valid properties
1230 - 'party_a' - make this channel the preferred Party A in any CDR between two
1231 channels. If two channels have this property set, the creation time of the
1232 channel is used to determine who is Party A. Note that dialed channels are
1233 never Party A in a CDR.
1234 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1235 application when set to True, and analogous to the 'e' option in ResetCDR
1240 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1241 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1242 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1245 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1246 string, i.e., [[context],extension],priority. If set on a channel, if a
1247 channel leaves a bridge but is not hung up it will resume dialplan execution
1252 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1253 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1254 The value of this setting is ignored when disabled is used for the argument.
1258 * A new function provided by chan_pjsip, this function can be used in
1259 conjunction with the Dial application to construct a dial string that will
1260 dial all contacts on an Address of Record associated with a chan_pjsip
1265 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1266 outbound channel prior to dialing.
1270 * Redirecting reasons can now be set to arbitrary strings. This means
1271 that the REDIRECTING dialplan function can be used to set the redirecting
1272 reason to any string. It also allows for custom strings to be read as the
1273 redirecting reason from SIP Diversion headers.
1277 * The SPEECH_ENGINE function now supports read operations. When read from, it
1278 will return the current value of the requested attribute.
1282 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1283 system as mailbox@context. The rest of the system cannot add @default
1284 to mailbox identifiers for app_voicemail that do not specify a context
1285 any longer. It is a mailbox identifier format that should only be
1286 interpreted by app_voicemail.
1292 res_agi (Asterisk Gateway Interface)
1294 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1296 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1299 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1300 will start the playback of the audio at the position specified. It will
1301 also return the final position of the file in 'endpos'.
1303 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1304 channel variable if the user stopped the file playback or if a remote
1305 entity stopped the playback. If neither stopped the playback, it will
1306 indicate the overall success/failure of the playback. If stopped early,
1307 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1310 * The SAY ALPHA command now accepts an additional parameter to control
1311 whether it specifies the case of uppercase, lowercase, or all letters to
1312 provide functionality similar to SayAlphaCase.
1314 res_ari (Asterisk RESTful Interface) (and others)
1316 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1317 control telephony primitives in Asterisk by remote client. This includes
1318 channels, bridges, endpoints, media, and other fundamental concepts. Users
1319 of ARI can develop their own communications applications, controlling
1320 multiple channels using an HTTP RESTful interface and receiving JSON events
1321 about the objects via a WebSocket connection. ARI can be configured in
1322 Asterisk via ari.conf. For more information on ARI, see
1323 https://wiki.asterisk.org/wiki/x/0YCLAQ
1327 * Parking has been extracted from the Asterisk core as a loadable module,
1328 res_parking. Configuration for parking is now provided by res_parking.conf.
1329 Configuration through features.conf is no longer supported.
1331 * res_parking uses the configuration framework. If an invalid configuration is
1332 supplied, res_parking will fail to load or fail to reload. Previously,
1333 invalid configurations would generally be accepted, with certain errors
1334 resulting in individually disabled parking lots.
1336 * Parked calls are now placed in bridges. While this is largely an
1337 architectural change, it does have implications on how channels in a parking
1338 lot are viewed. For example, commands that display channels in bridges will
1339 now also display the channels in a parking lot.
1341 * The order of arguments for the new parking applications have been modified.
1342 Timeout and return context/exten/priority are now implemented as options,
1343 while the name of the parking lot is now the first parameter. See the
1344 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1345 in-depth information as well as syntax.
1347 * Extensions are by default no longer automatically created in the dialplan to
1348 park calls or pickup parked calls. Generation of dialplan extensions can be
1349 enabled using the 'parkext' configuration option.
1351 * ADSI functionality for parking is no longer supported. The 'adsipark'
1352 configuration option has been removed as a result.
1354 * The PARKINGSLOT channel variable has been deprecated in favor of
1355 PARKING_SPACE to match the naming scheme of the new system.
1357 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1358 channel even when the configuration option 'comebactoorigin' is enabled.
1360 * A new CLI command 'parking show' has been added. This allows a user to
1361 inspect the parking lots that are currently in use.
1362 'parking show <parkinglot>' will also show the parked calls in a specific
1365 * The CLI command 'parkedcalls' is now deprecated in favor of
1366 'parking show <parkinglot>'.
1368 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1369 can be used to get a list of parked calls for a specific parking lot.
1371 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1372 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1373 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1374 longer a required argument.
1376 * The ParkAndAnnounce application is now provided through res_parking instead
1377 of through the separate app_parkandannounce module.
1379 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1380 by default. Instead, it will follow the timeout rules of the parking lot. The
1381 old behavior can be reproduced by using the 'c' option.
1383 * Dynamic parking lots will now fail to be created under the following
1385 - if the parking lot specified by PARKINGDYNAMIC does not exist
1386 - if they require exclusive park and parkedcall extensions which overlap
1387 with existing parking lots.
1389 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1390 currently contain no calls. Dynamic parking lots containing parked calls
1391 will persist through the reloads without alteration.
1393 * If 'parkext_exclusive' is set for a parking lot and that extension is
1394 already in use when that parking lot tries to register it, this is now
1395 considered a parking system configuration error. Configurations which do
1396 this will be rejected.
1398 * Added channel variable PARKER_FLAT. This contains the name of the extension
1399 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1400 comebacktoorigin is disabled, but the dialplan or an external control
1401 mechanism wants to use the extension in the park-dial context that was
1402 generated to re-dial the parker on timeout.
1404 res_pjsip (and many others)
1406 * A large number of resource modules make up the SIP stack based on pjsip.
1407 The chan_pjsip channel driver users these resource modules to provide
1408 various SIP functionality in Asterisk. The majority of configuration for
1409 these modules is performed in pjsip.conf. Other modules may use their
1410 own configuration files.
1412 * Added 'set_var' option for an endpoint. For each variable specified that
1413 variable gets set upon creation of a channel involving the endpoint.
1417 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1418 them, an Asterisk-specific version of PJSIP needs to be installed.
1419 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1421 res_statsd/res_chan_stats
1423 * A new resource module, res_statsd, has been added, which acts as a statsd
1424 client. This module allows Asterisk to publish statistics to a statsd
1425 server. In conjunction with res_chan_stats, it will publish statistics about
1426 channels to the statsd server. It can be configured via res_statsd.conf.
1430 * Device state for XMPP buddies is now available using the following format:
1431 XMPP/<client name>/<buddy address>
1432 If any resource is available the device state is considered to be not in use.
1433 If no resources exist or all are unavailable the device state is considered
1440 Realtime/Database Scripts
1442 * Asterisk previously included example db schemas in the contrib/realtime/
1443 directory of the source tree. This has been replaced by a set of database
1444 migrations using the Alembic framework. This allows you to use alembic to
1445 initialize the database for you. It will also serve as a database migration
1446 tool when upgrading Asterisk in the future.
1448 See contrib/ast-db-manage/README.md for more details.
1452 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1453 This python script will convert an existing sip.conf file to a
1454 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1455 is meant to be an aid in converting an existing chan_sip configuration to
1456 a chan_pjsip configuration, but it is expected that configuration beyond
1457 what the script provides will be needed.
1460 ------------------------------------------------------------------------------
1461 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1462 ------------------------------------------------------------------------------
1466 * The Asterisk build system will now build and install a shared library
1467 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1468 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1469 that Asterisk can ensure that these functions do *not* get called by any
1470 modules that are loaded into Asterisk, since they should only be called once
1471 in any single process. If desired, this feature can be disabled by supplying
1472 the "--disable-asteriskssl" option to the configure script.
1474 * A new make target, 'full', has been added to the Makefile. This performs
1475 the same compilation actions as make all, but will also scan the entirety of
1476 each source file for documentation. This option is needed to generate AMI
1477 event documentation. Note that your system must have Python in order for
1478 this make target to succeed.
1480 * The optimization portion of the build system has been reworked to avoid
1481 broken builds on certain architectures. All architecture-specific
1482 optimization has been removed in favor of using -march=native to allow gcc
1483 to detect the environment in which it is running when possible. This can
1484 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1486 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1487 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1489 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1490 previously parsed the header file to obtain the version of Asterisk, you
1491 will now have to go through Asterisk to get the version information.
1499 * Added 'F()' option. Similar to the dial option, this can be supplied with
1500 arguments indicating where the callee should go after the caller is hung up,
1501 or without options specified, the priority after the Queue will be used.
1506 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1507 all non-admin participants on a conference. The confbridge configuration
1508 file also allows for the default sounds played to all conference users when
1509 this occurs to be overriden using sound_participants_unmuted and
1510 sound_participants_muted.
1512 * Added menu action participant_count. This will playback the number of
1513 current participants in a conference.
1515 * Added announcement configuration option to user profile. If set the sound
1516 file will be played to the user, and only the user, upon joining the
1519 * Added record_file_append option that defaults to "yes", but if set to no
1520 will create a new file between each start/stop recording.
1525 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1526 channels respectively before the callee channels are called.
1531 * Added support for IPv6.
1533 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1534 external process will cause the current playlist to be cleared, including
1535 stopping any audio file that is currently playing. This is useful when you
1536 want to interrupt audio playback only when specific DTMF is entered by the
1542 * A new option, 'I' has been added to app_followme. By setting this option,
1543 Asterisk will not update the caller with connected line changes when they
1544 occur. This is similar to app_dial and app_queue.
1546 * The 'N' option is now ignored if the call is already answered.
1548 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1549 and caller channels respectively before the callee channels are called.
1551 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1557 * MixMonitor hooks now have IDs associated with them which can be used to
1558 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1559 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1560 now accepts that ID as an argument.
1562 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1563 indicated mailboxes.
1568 * The connect action in app_mysql now allows you to specify a port number to
1569 connect to. This is useful if you run a MySQL server on a non-standard
1575 * Increased the default number of allowed destinations from 5 to 12.
1580 * The app_page application now no longer depends on DAHDI or app_meetme. It
1581 has been re-architected to use app_confbridge internally.
1586 * Added queue options autopausebusy and autopauseunavail for automatically
1587 pausing a queue member when their device reports busy or congestion.
1589 * The 'ignorebusy' option for queue members has been deprecated in favor of
1590 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1591 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1592 per interface basis. Individual ringinuse values can now be set in
1593 queues.conf via an argument to member definitions. Lastly, the queue
1594 'ringinuse' setting now only determines defaults for the per member
1595 'ringinuse' setting and does not override per member settings like it does
1596 in earlier versions.
1598 * Added 'F()' option. Similar to the dial option, this can be supplied with
1599 arguments indicating where the callee should go after the caller is hung up,
1600 or without options specified, the priority after the Queue will be used.
1602 * Added new option log_member_name_as_agent, which will cause the membername to
1603 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1604 state_interface has been set.
1606 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1608 * App_queue will now play periodic announcements for the caller that
1609 holds the first position in the queue while waiting for answer.
1613 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1614 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1615 changed arguments to SayUnixTime so that every option is truly optional even
1616 when using multiple options (so that j option could be used without having to
1617 manually specify timezone and format) There are other benefits, e.g., format
1618 can now be used without specifying time zone as well.
1623 * Addition of the VM_INFO function - see Function changes.
1625 * The imapserver, imapport, and imapflags configuration options can now be
1626 overriden on a user by user basis.
1628 * When voicemail plays a message's envelope with saycid set to yes, when
1629 reaching the caller id field it will play a recording of a file with the same
1630 base name as the sender's callerid if there is a similarly named file in
1631 <astspooldir>/recordings/callerids/
1633 * Voicemails now contains a unique message identifier "msg_id", which is stored
1634 in the message envelope with the sound files. IMAP backends will now store
1635 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1636 backends will store the message identifier in a "msg_id" column. See
1637 UPGRADE.txt for more information.
1639 * Added VoiceMailPlayMsg application. This application will play a single
1640 voicemail message from a mailbox. The result of the application, SUCCESS or
1641 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1646 * Hangup handlers can be attached to channels using the CHANNEL() function.
1647 Hangup handlers will run when the channel is hung up similar to the h
1648 extension. The hangup_handler_push option will push a GoSub compatible
1649 location in the dialplan onto the channel's hangup handler stack. The
1650 hangup_handler_pop option will remove the last added location, and optionally
1651 replace it with a new GoSub compatible location. The hangup_handler_wipe
1652 option will remove all locations on the stack, and optionally add a new
1655 * The expression parser now recognizes the ABS() absolute value function,
1656 which will convert negative floating point values to positive values.
1658 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1659 control of faxdetect.
1661 * Addition of the VM_INFO function that can be used to retrieve voicemail
1662 user information, such as the email address and full name.
1663 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1666 * The REDIRECTING function now supports the redirecting original party id
1669 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1670 lets you set some of the configuration options from the [general] section
1671 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1672 the key sequence used to activate built-in features, such as blindxfer,
1673 and automon. See the built-in documentation for details.
1675 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1676 instead of simply the uri. This is the format that MessageSend() can use
1677 in the from parameter for outgoing SIP messages.
1679 * Added the PRESENCE_STATE function. This allows retrieving presence state
1680 information from any presence state provider. It also allows setting
1681 presence state information from a CustomPresence presence state provider.
1682 See AMI/CLI changes for related commands.
1684 * Added the AMI_CLIENT function to make manager account attributes available
1685 to the dialplan. It currently supports returning the current number of
1686 active sessions for a given account.
1688 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1689 and the REDIRECTING functions.
1697 * Added a manager event "LocalBridge" for local channel call bridges between
1698 the two pseudo-channels created.
1703 * Added dialtone_detect option for analog ports to disconnect incoming
1704 calls when dialtone is detected.
1706 * Added option colp_send to send ISDN connected line information. Allowed
1707 settings are block, to not send any connected line information; connect, to
1708 send connected line information on initial connect; and update, to send
1709 information on any update during a call. Default is update.
1711 * Add options namedcallgroup and namedpickupgroup to support installations
1712 where a higher number of groups (>64) is required.
1714 * Added support to use private party ID information with PRI calls.
1719 * A new channel driver named chan_motif has been added which provides support for
1720 Google Talk and Jingle in a single channel driver. This new channel driver includes
1721 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1722 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1723 specification, current Google Jingle specification, and the original Google Talk
1729 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1730 globally and overriden on a peer by peer basis.
1732 * Direct media functionality has been added. Options in config are:
1733 directmedia (directrtp) and directrtpsetup (earlydirect)
1735 * ChannelUpdate events now contain a CallRef header.
1740 * Asterisk will no longer substitute CID number for CID name in the display
1741 name field if CID number exists without a CID name. This change improves
1742 compatibility with certain device features such as Avaya IP500's directory
1745 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1746 created using that setting to not be removed during SIP reload.
1748 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1749 request with a "Record:" header, this will turn the requested feature on/off.
1750 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1751 dynamic features must be enabled and configured properly on the requesting
1752 channel for this to function properly.
1754 * Add support to realtime for the 'callbackextension' option.
1756 * When multiple peers exist with the same address, but differing
1757 callbackextension options, incoming requests that are matched by address
1758 will be matched to the peer with the matching callbackextension if it is
1761 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1762 which set the force_rport and comedia options automatically if Asterisk
1763 detects that an incoming SIP request crossed a NAT after being sent by
1764 the remote endpoint.
1766 * The default global nat setting in sip.conf has been changed from force_rport
1767 to auto_force_rport.
1769 * NAT settings are now a combinable list of options. The equivalent of the
1770 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1772 * Adds an option send_diversion which can be disabled to prevent
1773 diversion headers from automatically being added to INVITE requests.
1775 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1776 be sent to the remote host at a given interval to keep the NAT mapping open.
1777 This can be enabled using the keepalive configuration option.
1779 * Add option 'tonezone' to specify country code for indications. This option
1780 can be set both globally and overridden for specific peers.
1782 * The SIP Security Events Framework now supports IPv6.
1784 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1785 between multiple user agents. When set, for directmedia reinvites,
1786 Asterisk will not send an immediate reinvite on an incoming call leg. This
1787 option is useful when peered with another SIP user agent that is known to
1788 send immediate direct media reinvites upon call establishment.
1790 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1793 * Add options subminexpiry and submaxexpiry to set limits of subscription
1794 timer independently from registration timer settings. The setting of the
1795 registration timer limits still is done by options minexpiry, maxexpiry
1796 and defaultexpiry. For backwards compatibility the setting of minexpiry
1797 and maxexpiry also is used to configure the subscription timer limits if
1798 subminexpiry and submaxexpiry are not set in sip.conf.
1800 * Set registration timer limits to default values when reloading sip
1801 configuration and values are not set by configuration.
1803 * Add options namedcallgroup and namedpickupgroup to support installations
1804 where a higher number of groups (>64) is required.
1806 * When a MESSAGE request is received, the address the request was received from
1807 is now saved in the SIP_RECVADDR variable.
1809 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1810 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1811 the ANI2/OLI information is set on the channel, which can be retrieved using
1812 the CALLERID function.
1814 * Peers can now be configured to support negotiation of ICE candidates using
1815 the setting icesupport. See res_rtp_asterisk changes for more information.
1817 * Added support for format attribute negotiation. See the Codecs changes for
1820 * Extra headers specified with SIPAddHeader are sent with the REFER message
1821 when using Transfer application. See refer_addheaders in sip.conf.sample.
1823 * Added support to use private party ID information with calls.
1825 * Adds an option discard_remote_hold_retrieval that when set stops telling
1826 the peer to start music on hold.
1831 * Added skinny version 17 protocol support.
1835 --------------------
1836 * Added ability to use multiple lines for a single phone. This allows multiple
1837 calls to occur on a single phone, using callwaiting and switching between calls.
1839 * Added option 'sharpdial' allowing end dialing by pressing # key
1841 * Added option 'interdigit_timer' to control phone dial timeout
1843 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1845 * Added global 'debug' option, that enables debug in channel driver
1847 * Added ability to translate on-screen menu in multiple languages. Tested on
1848 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1849 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1852 * In addition to English added French and Russian languages for on-screen menus
1854 * Reworked dialing number input: added dialing by timeout, immediate dial on
1855 on dialplan compare, phone number length now not limited by screen size
1857 * Added ability to pickup a call using features.conf defined value and
1863 * Add options namedcallgroup and namedpickupgroup to support installations
1864 where a higher number of groups (>64) is required.
1866 * Added support to use private party ID information with calls.
1871 * The minimum DTMF duration can now be configured in asterisk.conf
1872 as "mindtmfduration". The default value is (as before) set to 80 ms.
1873 (previously it was only available in source code)
1875 * Named ACLs can now be specified in acl.conf and used in configurations that
1876 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1877 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1878 working ACL. In addition, some CLI commands have been added to provide
1879 show information and allow for module reloading - see CLI Changes.
1881 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1882 items (separated by commas), and items in the rule can be negated by prefixing
1883 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1884 longer necessray to control the order that the 'permit' and 'deny' columns are
1885 returned from queries.
1887 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1888 be used within the dynamic weight attribute when specifying a mapping.
1890 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1891 header, instead of putting the user defined event name there. When enabled
1892 the UserDefType header is added for user defined events. This feature is
1893 enabled with the setting show_user_defined.
1895 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1896 line purposes use the following variables instead of their macro equivalents:
1897 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1898 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1899 cc_callback_macro in channel configurations.
1901 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1904 * Call files now support the "early_media" option to connect with an outgoing
1905 extension when early media is received.
1907 * Added support to use private party ID information with calls.
1912 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1913 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1914 AGI application would exit immediately after a channel hangup is detected.
1916 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1917 are resolved and each address is attempted in turn until one succeeds or
1921 AMI (Asterisk Manager Interface)
1923 * The originate action now has an option "EarlyMedia" that enables the
1924 call to bridge when we get early media in the call. Previously,
1925 early media was disregarded always when originating calls using AMI.
1927 * Added setvar= option to manager accounts (much like sip.conf)
1929 * Originate now generates an error response if the extension given is not found
1932 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1933 them if the i(variable) option is used. StopMixMonitor will accept
1934 MixMonitorID as an option to close specific MixMonitors.
1936 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1937 updated to include information about peers configured with
1938 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1939 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1940 returned if auto_force_rport is not enabled.
1942 * Added SIPpeerstatus manager command which will generate PeerStatus events
1943 similar to the existing PeerStatus events found in chan_sip on demand.
1945 * Hangup now can take a regular expression as the Channel option. If you want
1946 to hangup multiple channels, use /regex/ as the Channel option. Existing
1947 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1948 the manager will send you a list of channels back that were hung up.
1950 * Support for IPv6 addresses has been added.
1952 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1953 documentation is only generated when Asterisk is compiled using 'make full'.
1954 See the CLI section for commands to display AMI event information.
1956 * The AMI Hangup event now includes the AccountCode header so you can easily
1957 correlate with AMI Newchannel events.
1959 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1960 the StateInterface of the queue member.
1962 * Added AMI event SessionTimeout in the Call category that is issued when a
1963 call is terminated due to either RTP stream inactivity or SIP session timer
1966 * CEL events can now contain a user defined header UserDefType. See core
1967 changes for more information.
1969 * OOH323 ChannelUpdate events now contain a CallRef header.
1971 * Added PresenceState command. This command will report the presence state for
1972 the given presence provider.
1974 * Added Parkinglots command. This will list all parking lots as a series of
1975 AMI Parkinglot events.
1977 * Added MessageSend command. This behaves in the same manner as the
1978 MessageSend application, and is a technolgoy agnostic mechanism to send out
1979 of call text messages.
1981 * Added "message" class authorization. This grants an account permission to
1982 send out of call messages. Write-only.
1987 * The "dialplan add include" command has been modified to create context a context
1988 if one does not already exist. For instance, "dialplan add include foo into bar"
1989 will create context "bar" if it does not already exist.
1991 * A "dialplan remove context" command has been added to remove a context from
1994 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1995 filenames of all running mixmonitors on a channel.
1997 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1998 numeric instead of 0, 1, or 2.
2000 * "stun show status" will show a table describing how the STUN client is
2003 * "acl show [named acl]" will show information regarding a Named ACL. The
2004 acl module can be reloaded with "reload acl".
2006 * Added CLI command to display AMI event information - "manager show events",
2007 which shows a list of all known and documented AMI events, and "manager show
2008 event [event name]", which shows detail information about a specific AMI
2011 * The result of the CLI command "queue show" now includes the state interface
2012 information of the queue member.
2014 * The command "core set verbose" will now set a separate level of logging for
2015 each remote console without affecting any other console.
2017 * Added command "cdr show pgsql status" to check connection status
2019 * "sip show channel" will now display the complete route set.
2021 * Added "presencestate list" command. This command will list all custom
2022 presence states that have been set by using the PRESENCE_STATE dialplan
2025 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2026 command. This changes a custom presence to a new state.
2031 * Codec lists may now be modified by the '!' character, to allow succinct
2032 specification of a list of codecs allowed and disallowed, without the
2033 requirement to use two different keywords. For example, to specify all
2034 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2036 * Add support for parsing SDP attributes, generating SDP attributes, and
2037 passing it through. This support includes codecs such as H.263, H.264, SILK,
2038 and CELT. You are able to set up a call and have attribute information pass.
2039 This should help considerably with video calls.
2041 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2042 just like the GSM codec.
2046 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2047 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2051 * Asterisk version and build information is now logged at the beginning of a
2054 * Threads belonging to a particular call are now linked with callids which get
2055 added to any log messages produced by those threads. Log messages can now be
2056 easily identified as involved with a certain call by looking at their call id.
2057 Call ids may also be attached to log messages for just about any case where
2058 it can be determined to be related to a particular call.
2060 * Each logging destination and console now have an independent notion of the
2061 current verbosity level. Logger.conf now allows an optional argument to
2062 the 'verbose' specifier, indicating the level of verbosity sent to that
2063 particular logging destination. Additionally, remote consoles now each
2064 have their own verbosity level. The command 'core set verbose' will now set
2065 a separate level for each remote console without affecting any other
2071 * Added 'announcement' option which will play at the start of MOH and between
2072 songs in modes of MOH that can detect transitions between songs (eg.
2078 * New per parking lot options: comebackcontext and comebackdialtime. See
2079 configs/features.conf.sample for more details.
2081 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2084 * Channel variable PARKEDCALL is now set with the name of the parking lot
2085 when a timeout occurs.
2091 CDR Postgresql Driver
2093 * Added command "cdr show pgsql status" to check connection status
2096 CDR Adaptive ODBC Driver
2098 * Added schema option for databases that support specifying a schema.
2106 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2107 CALENDAR_WRITE has completed successfully.
2112 * A new option, 'probation' has been added to rtp.conf
2113 RTP in strictrtp mode can now require more than 1 packet to exit learning
2114 mode with a new source (and by default requires 4). The probation option
2115 allows the user to change the required number of packets in sequence to any
2116 desired value. Use a value of 1 to essentially restore the old behavior.
2117 Also, with strictrtp on, Asterisk will now drop all packets until learning
2118 mode has successfully exited. These changes are based on how pjmedia handles
2119 media sources and source changes.
2121 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2122 enabled or disabled using the icesupport setting. A variety of other
2123 settings have been introduced to configure STUN/TURN connections.
2128 * A new module, res_corosync, has been introduced. This module uses the
2129 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2130 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2131 Device State (presence) information. This module is very similar to, and
2132 is a replacement for the res_ais module that was in previous releases of
2138 * This module adds a cleaned up, drop-in replacement for res_jabber called
2139 res_xmpp. This provides the same externally facing functionality but is
2140 implemented differently internally. res_jabber has been deprecated in favor
2141 of res_xmpp; please see the UPGRADE.txt file for more information.
2146 * The safe_asterisk script has been updated to allow several of its parameters
2147 to be set from environment variables. This also enables a custom run
2148 directory of Asterisk to be specified, instead of defaulting to /tmp.
2150 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2151 its value to determine the directory to assume is the top-level directory of
2152 the source tree. If the variable is not set, it defaults to the current
2153 behavior and uses the current working directory.
2155 ------------------------------------------------------------------------------
2156 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2157 ------------------------------------------------------------------------------
2161 * Asterisk now has protocol independent support for processing text messages
2162 outside of a call. Messages are routed through the Asterisk dialplan.
2163 SIP MESSAGE and XMPP are currently supported. There are options in
2164 jabber.conf and sip.conf to allow enabling these features.
2165 -> jabber.conf: see the "sendtodialplan" and "context" options.
2166 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2167 and "outofcall_message_context" options.
2168 The MESSAGE() dialplan function and MessageSend() application have been
2169 added to go along with this functionality. More detailed usage information
2170 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2171 * If real-time text support (T.140) is negotiated, it will be preferred for
2172 sending text via the SendText application. For example, via SIP, messages
2173 that were once sent via the SIP MESSAGE request would be sent via RTP if
2174 T.140 text is negotiated for a call.
2178 * parkedmusicclass can now be set for non-default parking lots.
2180 Asterisk Manager Interface
2181 --------------------------
2182 * PeerStatus now includes Address and Port.
2183 * Added Hold events for when the remote party puts the call on and off hold
2184 for chan_dahdi ISDN channels.
2185 * Added new action MeetmeListRooms to list active conferences (shows same
2186 data as "meetme list" at the CLI).
2187 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2188 Description field that is set by 'description' in the channel configuration
2190 * Added Uniqueid header to UserEvent.
2191 * Added new action FilterAdd to control event filters for the current session.
2192 This requires the system permission and uses the same filter syntax as
2193 filters that can be defined in manager.conf
2194 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2195 versions had some instances of the event converted, but others were left
2196 as-is. All Unlink events should now be converted to Bridge events. The AMI
2197 protocol version number was incremented to 1.2 as a result of this change.
2199 Asterisk HTTP Server
2200 --------------------------
2201 * The HTTP Server can bind to IPv6 addresses.
2204 --------------------------
2205 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2206 with busydetect. usage example: busypattern=200,200,200,600
2209 --------------------------
2210 * New 'gtalk show settings' command showing the current settings loaded from
2212 * The 'logger reload' command now supports an optional argument, specifying an
2213 alternate configuration file to use.
2214 * 'dialplan add extension' command will now automatically create a context if
2215 the specified context does not exist with a message indicated it did so.
2216 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2217 Description field which can be populated with 'description' in the channel
2218 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2221 --------------------------
2222 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2223 thus allowing records which do NOT match the specified filter.
2224 * Added ability to log CONGESTION calls to CDR
2227 --------------------------
2228 * Ability to define custom SILK formats in codecs.conf.
2229 * Addition of speex32 audio format with translation.
2230 * CELT codec pass-through support and ability to define
2231 custom CELT formats in codecs.conf.
2232 * Ability to read raw signed linear files with sample rates
2233 ranging from 8khz - 192khz. The new file extensions introduced
2234 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2235 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2236 Skinny, H.323, etc) can still only support the following codecs:
2237 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2238 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2239 Video: h261, h263, h263p, h264, mpeg4
2244 --------------------------
2245 * New highly optimized and customizable ConfBridge application capable of
2246 mixing audio at sample rates ranging from 8khz-96khz.
2247 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2248 and bridge profiles on a channel.
2249 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2250 about a conference such as locked status and number of parties, admins,
2252 * Addition of video_mode option in confbridge.conf for adding video support
2253 into a bridge profile.
2254 * Addition of the follow_talker video_mode in confbridge.conf. This video
2255 mode dynamically switches the video feed to always display the loudest talker
2256 supplying video in the conference.
2260 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2261 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2262 variables from asterisk.conf.
2266 * Addition of the JITTERBUFFER dialplan function. This function allows
2267 for jitterbuffering to occur on the read side of a channel. By using
2268 this function conference applications such as ConfBridge and MeetMe can
2269 have the rx streams jitterbuffered before conference mixing occurs.
2270 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2272 * Added STRREPLACE function. This function let's the user search a variable
2273 for a given string to replace with another string as many times as the
2274 user specifies or just throughout the whole string.
2275 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2276 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2277 * Added extensions to chan_ooh323 in function CHANNEL()
2279 libpri channel driver (chan_dahdi) DAHDI changes
2280 --------------------------
2281 * Added moh_signaling option to specify what to do when the channel's bridged
2282 peer puts the ISDN channel on hold.
2283 * Added display_send and display_receive options to control how the display ie
2284 is handled. To send display text from the dialplan use the SendText()
2285 application when the option is enabled.
2286 * Added mcid_send option to allow sending a MCID request on a span.
2289 --------------------------
2290 * Added setvar option to calendar.conf to allow setting channel variables on
2291 notification channels.
2292 * Added "calendar show types" CLI command to list registered calendar
2296 --------------------------
2297 * Added two new options, r and t with file name arguments to record
2298 single direction (unmixed) audio recording separate from the bidirectional
2299 (mixed) recording. The mixed file name argument is optional now as long
2300 as at least one recording option is used.
2303 --------------------------
2304 * Added a new option, l, which will disable local call optimization for
2305 channels involved with the FollowMe thread. Use this option to improve
2306 compatability for a FollowMe call with certain dialplan apps, options, and
2310 --------------------------
2311 * Added option "k" that will automatically close the conference when there's
2312 only one person left when a user exits the conference.
2315 --------------------------
2316 * cel_pgsql now supports the 'extra' column for data added using the
2317 CELGenUserEvent() application.
2320 --------------------------
2321 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2322 in the sample extensions.lua file for syntax details.
2323 * Applications that perform jumps in the dialplan such as Goto will now
2324 execute properly. When pbx_lua detects that the context, extension, or
2325 priority we are executing on has changed it will immediately return control
2326 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2327 the priority after the currently executing priority.
2328 * An autoservice is now started by default for pbx_lua channels. It can be
2329 stopped and restarted using the autoservice_stop() and autoservice_start()
2333 --------------------------
2334 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2335 into a FAXStatus event with an 'Operation' header that will be either
2336 'send', 'receive', and 'gateway'.
2337 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2338 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2339 feature will handle converting a fax call between an audio T.30 fax terminal
2340 and an IFP T.38 fax terminal.
2344 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2345 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2346 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2350 * Added general option negative_penalty_invalid default off. when set
2351 members are seen as invalid/logged out when there penalty is negative.
2352 for realtime members when set remove from queue will set penalty to -1.
2353 * Added queue option autopausedelay when autopause is enabled it will be
2354 delayed for this number of seconds since last successful call if there
2355 was no prior call the agent will be autopaused immediately.
2356 * Added member option ignorebusy this when set and ringinuse is not
2357 will allow per member control of multiple calls as ringinuse does for
2362 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2364 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2365 one participant left (much like a normal call bridge)
2366 * Added extra argument to Originate to set timeout.
2370 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2371 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2372 utility in the UTILS section of menuselect. If an existing astdb is found and no
2373 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2374 convert an existing astdb to the SQLite3 version automatically at runtime.
2378 * Modules marked as deprecated are no longer marked as building by default. Enabling
2379 these modules is still available via menuselect.
2383 * authdebug is now disabled by default. To enable this functionaility again
2384 set authdebug = yes in iax.conf.
2388 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2389 releases it was disabled.
2393 * The PBX core previously made a call with a non-existing extension test for
2394 extension s@default and jump there if the extension existed.
2395 This was a bad default behaviour and violated the principle of least surprise.
2396 It has therefore been changed in this release. It may affect some
2397 applications and configurations that rely on this behaviour. Most channel
2398 drivers have avoided this for many releases by testing whether the extension
2399 called exists before starting the PBX and generating a local error.
2400 This behaviour still exists and works as before.
2402 Extension "s" is used when no extension is given in a channel driver,
2403 like immediate answer in DAHDI or calling to a domain with no user part
2406 ------------------------------------------------------------------------------
2407 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2408 ------------------------------------------------------------------------------
2412 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2413 now defaults to force_rport. It is very important that phones requiring nat=no be
2414 specifically set as such instead of relying on the default setting. If at all
2415 possible, all devices should have nat settings configured in the general section as
2416 opposed to configuring nat per-device.
2417 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2418 codecs sent in response to an INVITE to the single most preferred codec.
2419 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2420 to be used for the outgoing call. It must be one of the codecs configured
2422 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2423 to be used for holding a private key. If tlsprivatekey is not specified,
2424 tlscertfile is searched for both public and private key.
2425 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2426 outbound client connections to be specified.
2427 * The sendrpid parameter has been expanded to include the options
2428 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2429 header to be sent (equivalent to setting sendrpid=yes) and setting
2430 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2431 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2432 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2433 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2434 will accept the SDP even if the SDP version number is not properly incremented,
2435 but will generate a warning in the log indicating that the SIP peer that sent
2436 the SDP should have the 'ignoresdpversion' option set.
2437 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2438 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2439 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2440 remote side requests it and disables symmetric RTP support. Setting it to
2441 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2442 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2443 and enables symmetric RTP support.
2444 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2445 response. This permits the master channel to know how each channel dialled
2446 in a multi-channel setup resolved in an individual way. This carries a
2447 performance penalty and can be disabled in sip.conf using the
2448 'storesipcause' option.
2449 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2450 configuration for the externip and externhost options when tcp or tls is used.
2451 * Added support for message body (stored in content variable) to SIP NOTIFY message
2452 accessible via AMI and CLI.
2453 * Added 'media_address' configuration option which can be used to explicitly specify
2454 the IP address to use in the SDP for media (audio, video, and text) streams.
2455 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2456 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2458 * Added 'use_q850_reason' configuration option for generating and parsing
2459 if available Reason: Q.850;cause=<cause code> header. It is implemented
2460 in some gateways for better passing PRI/SS7 cause codes via SIP.
2461 * When dialing SIP peers, a new component may be added to the end of the dialstring
2462 to indicate that a specific remote IP address or host should be used when dialing
2463 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2464 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2465 ability to selectively force bridged channels to also be encrypted is also
2466 implemented. Branching in the dialplan can be done based on whether or not
2467 a channel has secure media and/or signaling.
2468 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2470 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2471 Charge messages to snom phones.
2472 * Added support for G.719 media streams.
2473 * Added support for 16khz signed linear media streams.
2474 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2475 RTP has been outfitted with the same abilities.
2476 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2477 available in device configurations as well as in the dial plan.
2478 * Addition of the 'subscribe_network_change' option for turning on and off
2479 res_stun_monitor module support in chan_sip.
2480 * Addition of the 'auth_options_requests' option for turning on and off
2481 authentication for OPTIONS requests in chan_sip.
2485 * Add #tryinclude statement for config files. This provides the same
2486 functionality as the #include statement however an asterisk module will
2487 still load if the filename does not exist. Using the #include statement
2488 Asterisk will not allow the module to load.
2492 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2493 on realtime updates.
2494 * Added the ability for chan_iax2 to inform the dialplan whether or not
2495 encryption is being used. This interoperates with the SIP SRTP implementation
2496 so that a secure SIP call can be bridged to a secure IAX call when the
2497 dialplan requires bridged channels to be "secure".
2498 * Addition of the 'subscribe_network_change' option for turning on and off
2499 res_stun_monitor module support in chan_iax.
2504 * Added ability to preset channel variables on indicated lines with the setvar
2505 configuration option. Also, clearvars=all resets the list of variables back
2507 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2508 See configs/res_pktccops.conf for more information.
2510 XMPP Google Talk/Jingle changes
2511 -------------------------------
2512 * Added the externip option to gtalk.conf.
2513 * Added the stunaddr option to gtalk.conf which allows for the automatic
2514 retrieval of the external ip from a stun server.
2518 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2519 match to a partial channel name.
2520 * Added .m3u support for Mp3Player application.
2521 * Added progress option to the app_dial D() option. When progress DTMF is
2522 present, those values are sent immediately upon receiving a PROGRESS message
2523 regardless if the call has been answered or not.
2524 * Added functionality to the app_dial F() option to continue with execution
2525 at the current location when no parameters are provided.
2526 * Added the 'a' option to app_dial to answer the calling channel before any
2527 announcements or macros are executed.
2528 * Modified app_dial to set answertime when the called channel answers even if
2529 the called channel hangs up during playback of an announcement.
2530 * Modified app_dial 'r' option to support an additional parameter to play an
2531 indication tone from indications.conf
2532 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2533 to cycle through the next available channel. By default this is still '*'.
2534 * Added x() option to app_chanspy. This option allows DTMF to be set to
2535 exit the application.
2536 * The Voicemail application has been improved to automatically ignore messages
2537 that only contain silence.
2538 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2539 associated mailbox(es) to be greetings-only.
2540 * The ChanSpy application now has the 'S' option, which makes the application
2541 automatically exit once it hits a point where no more channels are available
2543 * The ChanSpy application also now has the 'E' option, which spies on a single
2544 channel and exits when that channel hangs up.
2545 * The MeetMe application now turns on the DENOISE() function by default, for
2546 each participant. In our tests, this has significantly decreased background
2547 noise (especially noisy data centers).
2548 * Voicemail now permits storage of secrets in a separate file, located in the
2549 spool directory of each individual user. The control for this is located in
2550 the "passwordlocation" option in voicemail.conf. Please see the sample
2551 configuration for more information.
2552 * The ChanIsAvail application now exposes the returned cause code using a separate
2553 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2554 * Added 'd' option to app_followme. This option disables the "Please hold"
2556 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2557 received will terminate recording.
2558 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2559 Previously the folder could only be set per context, but has now been extended
2560 using the imapfolder option.
2561 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2562 * Voicemail now allows the pager date format to be specified separately from the
2564 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2565 to allow joining, leaving, and sending text to group chats.
2566 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2567 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2568 to all paged phones (and optionally excluding the caller's one using the new
2569 option 'n') before the call is bridged.
2570 * The 'f' option to Dial has been augmented to take an optional argument. If no
2571 argument is provided, the 'f' option works as it always has. If an argument is
2572 provided, then the connected party information of all outgoing channels created
2573 during the Dial will be set to the argument passed to the 'f' option.
2574 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2576 * The OSP lookup application adds in/outbound network ID, optional security,
2577 number portability, QoS reporting, destination IP port, custom info and service
2579 * Added new application VMSayName that will play the recorded name of the voicemail
2580 user if it exists, otherwise will play the mailbox number.
2581 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2582 retrieve state for a particular bridge, where <name> is the conference name
2583 * app_directory now allows exiting at any time using the operator or pound key.
2584 * Voicemail now supports setting a locale per-mailbox.
2585 * Two new applications are provided for declining counting phrases in multiple
2586 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2588 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2590 * Voicemail now includes rdnis within msgXXXX.txt file.
2591 * ExternalIVR now supports IPv6 addresses.
2592 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2593 at https://wiki.asterisk.org/wiki/x/oQBB
2594 * ParkedCall and Park can now specify the parking lot to use.
2598 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2599 over SRV records associated with a specific service. From the CLI, type
2600 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2601 details on how these may be used.
2602 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2603 pitch of a channel's tx and rx audio streams.
2604 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2605 setting various connected line and redirecting party information.
2606 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2607 support ISDN subaddressing.
2608 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2609 * For DAHDI channels, the CHANNEL() dialplan function now allows
2610 the dialplan to request changes in the configuration of the active
2611 echo canceller on the channel (if any), for the current call only.
2614 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2616 The possible values are:
2618 on - normal mode (the echo canceller is actually reinitialized)
2620 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2622 voice - voice mode (returns from FAX mode, reverting the changes that
2623 were made when FAX mode was requested)
2624 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2625 and setting variables on the channel which created the current channel.
2626 Administrators should take care to avoid naming conflicts, when multiple
2627 channels are dialled at once, especially when used with the Local channel
2628 construct (which all could set variables on the master channel). Usage
2629 of the HASH() dialplan function, with the key set to the name of the slave
2630 channel, is one approach that will avoid conflicts.
2631 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2633 * func_odbc now allows multiple row results to be retrieved without using
2634 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2635 from the same query by using the name of the function which retrieved the
2636 first row as an argument to ODBC_FETCH().
2637 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2638 dialplan. This function returns the content of the received message.
2639 * Added REPLACE, which searches a given variable name for a set of characters,
2640 then either replaces them with a single character or deletes them.
2641 * Added PASSTHRU, which literally passes the same argument back as its return
2642 value. The intent is to be able to use a literal string argument to
2643 functions that currently require a variable name as an argument.
2644 * HASH-associated variables now can be inherited across channel creation, by
2645 prefixing the name of the hash at assignment with the appropriate number of
2646 underscores, just like variables.
2647 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2648 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2649 whether or not channels that are bridged to the current channel will be
2650 required to have secure signaling and/or media.
2651 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2652 the current channel has secure signaling and/or media.
2653 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2654 "no_media_path" option.
2655 Returns "0" if there is a B channel associated with the call.
2656 Returns "1" if no B channel is associated with the call. The call is either
2657 on hold or is a call waiting call.
2658 * Added option to dialplan function CDR(), the 'f' option
2659 allows for high resolution times for billsec and duration fields.
2660 * FILE() now supports line-mode and writing.
2661 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2662 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2666 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2667 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2668 and is set when a dynamic feature is triggered.
2669 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2670 to dynamically create a new parking lot matching the value this varible is
2672 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2673 features.conf that should be the base for dynamic parkinglots.
2674 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2675 parkinglot should have.
2676 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2677 parkinglot should have.
2678 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2683 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2684 timeout has expired.
2685 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2686 to the caller when an Agent's phone is ringing. This can be used to indicate
2687 to the caller that their call is about to be picked up, which is nice when
2688 one has been on hold for an extened period of time.
2689 * A new config option, penaltymemberslimit, has been added to queues.conf.
2690 When set this option will disregard penalty settings when a queue has too
2692 * A new option, 'I' has been added to both app_queue and app_dial.
2693 By setting this option, Asterisk will not update the caller with
2694 connected line changes or redirecting party changes when they occur.
2695 * A 'relative-periodic-announce' option has been added to queues.conf. When
2696 enabled, this option will cause periodic announce times to be calculated
2697 from the end of announcements rather than from the beginning.
2698 * The autopause option in queues.conf can be passed a new value, "all." The
2699 result is that if a member becomes auto-paused, he will be paused in all
2700 queues for which he is a member, not just the queue that failed to reach
2702 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2703 * The queue logger now allows events to optionally propagate to a file,
2704 even when realtime logging is turned on. Additionally, realtime logging
2705 supports sending the event arguments to 5 individual fields, although it
2706 will fallback to the previous data definition, if the new table layout is
2709 mISDN channel driver (chan_misdn) changes
2710 ----------------------------------------
2711 * Added display_connected parameter to misdn.conf to put a display string
2712 in the CONNECT message containing the connected name and/or number if
2713 the presentation setting permits it.
2714 * Added display_setup parameter to misdn.conf to put a display string
2715 in the SETUP message containing the caller name and/or number if the
2716 presentation setting permits it.
2717 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2718 indicate the dialplan settings are to be obtained from the asterisk
2720 * Made misdn.conf parameter callerid accept the "name" <number> format
2721 used by the rest of the system.
2722 * Made use the nationalprefix and internationalprefix misdn.conf
2723 parameters to prefix any received number from the ISDN link if that
2724 number has the corresponding Type-Of-Number. NOTE: This includes
2725 comparing the incoming call's dialed number against the MSN list.
2726 * Added the following new parameters: unknownprefix, netspecificprefix,
2727 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2728 received number from the ISDN link if that number has the corresponding
2730 * Added new dialplan application misdn_command which permits controlling
2731 the CCBS/CCNR functionality.
2732 * Added new dialplan function mISDN_CC which permits retrieval of various
2733 values from an active call completion record.
2734 * For PTP, you should manually send the COLR of the redirected-to party
2735 for an incomming redirected call if the incoming call could experience
2736 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2737 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2738 if the REDIRECTING(from-num) is not empty.
2739 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2740 option on all of the REDIRECTING statements before dialing the
2741 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2742 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2743 redirecting-to presentation (COLR) when it becomes available.
2744 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2747 thirdparty mISDN enhancements
2748 -----------------------------
2749 mISDN has been modified by Digium, Inc. to greatly expand facility message
2751 * Enhanced COLP support for call diversion and transfer.
2752 * CCBS/CCNR support.
2754 The latest modified mISDN v1.1.x based version is available at:
2755 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2756 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2758 Tagged versions of the modified mISDN code are available under:
2759 http://svn.digium.com/svn/thirdparty/mISDN/tags
2760 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2762 libpri channel driver (chan_dahdi) DAHDI changes
2763 -------------------------------------------
2764 * The channel variable PRIREDIRECTREASON is now just a status variable
2765 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2766 to read and alter the reason.
2767 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2768 redirected-to party for an incomming redirected call if the incoming call
2769 could experience further redirects. Just set the
2770 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2771 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2773 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2774 use the inhibit(i) option on all of the REDIRECTING statements before
2775 dialing the redirected-to party. You still have to set the
2776 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2777 will update the redirecting-to presentation (COLR) when it becomes available.
2778 * Added the ability to ignore calls that are not in a Multiple Subscriber
2779 Number (MSN) list for PTMP CPE interfaces.
2780 * Added dynamic range compression support for dahdi channels. It is
2781 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2782 * Added support for ISDN calling and called subaddress with partial support
2783 for connected line subaddress.
2784 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2785 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2786 to transfer a held call on disconnect similar to an analog phone.
2787 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2788 Will reroute/deflect an outgoing call when receive the message.
2789 Can use the DAHDISendCallreroutingFacility to send the message for the
2791 * Added standard location to add options to chan_dahdi dialing:
2792 Dial(DAHDI/g1[/extension[/options]])
2795 R Reverse charging indication
2796 * Added Reverse Charging Indication (Collect calls) send/receive option.
2797 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2798 Dial(DAHDI/g1/extension/R)
2799 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2800 (requires latest LibPRI)
2801 * Added ability to send/receive keypad digits in the SETUP message.
2802 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2803 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2804 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2805 (requires latest LibPRI)
2806 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2807 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2808 back into the same interface. Tromboned calls happen because of call routing,
2809 call deflection, call forwarding, and call transfer.
2810 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2811 * Added the ability to support call waiting calls. (The SETUP has no B channel
2813 * Added Malicious Call ID (MCID) event to the AMI call event class.
2814 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2816 Asterisk Manager Interface
2817 --------------------------
2818 * The Hangup action now accepts a Cause header which may be used to
2819 set the channel's hangup cause.
2820 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2821 to specify a separate .pem file to hold a private key. By default sslcert
2822 is used to hold both the public and private key.
2823 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2824 for options containing the 'tls' prefix. For example, 'sslenable' is now
2825 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2826 across all .conf files. All affected sample.conf files have been modified to
2827 reflect this change. Previous options such as 'sslenable' still work,
2828 but options with the 'tls' prefix are preferred.
2829 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2830 in a channel. (res_mutestream.so)
2831 * The configuration file manager.conf now supports a channelvars option, which
2832 specifies a list of channel variables to include in each channel-oriented
2834 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2835 and ExtraPriority to allow redirecting the second channel to a different
2836 location than the first.
2837 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2839 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2840 in a MixMonitor recording.
2841 * The 'iax2 show peers' output is now similar to the expected output of
2843 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2845 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2846 AOC-E messages on a channel.
2847 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2848 conform more closely to similar events.
2849 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2851 * Added optional parkinglot variable for park command.
2852 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2853 if CallerIDNum and CallerIDName headers are also present.
2855 Channel Event Logging
2856 ---------------------
2857 * A new interface, CEL, is introduced here. CEL logs single events, much like
2858 the AMI, but it differs from the AMI in that it logs to db backends much
2859 like CDR does; is based on the event subsystem introduced by Russell, and
2860 can share in all its benefits; allows multiple backends to operate like CDR;
2861 is specialized to event data that would be of concern to billing sytems,
2862 like CDR. Backends for logging and accounting calls have been produced,
2863 but a new CDR backend is still in development.
2867 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2868 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2869 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2870 * Multiple files and formats can now be specified in cdr_custom.conf.
2871 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2872 See configs/cdr_syslog.conf.sample for more information.
2873 * A 'sequence' field has been added to CDRs which can be combined with
2874 linkedid or uniqueid to uniquely identify a CDR.
2875 * Handling of billsec and duration field has changed. If your table definition
2876 specifies those fields as float,double or similar they will now be logged with
2877 microsecond accuracy instead of a whole integer.
2879 Calendaring for Asterisk
2880 ------------------------
2881 * A new set of modules were added supporing calendar integration with Asterisk.
2882 Dialplan functions for reading from and writing to calendars are included,
2883 as well as the ability to execute dialplan logic upon calendar event notifications.
2884 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2885 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2886 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2887 2003 support does not support forms-based authentication).
2889 Call Completion Supplementary Services for Asterisk
2890 ---------------------------------------------------
2891 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2892 DAHDI/ISDN supports call completion for the following switch types:
2893 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2894 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2896 Multicast RTP Support
2897 ---------------------
2898 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2899 The channel driver can be used with the Page application to perform multicast RTP
2900 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2901 Type can be either basic or linksys.
2902 Destination is the IP address and port for the RTP packets.
2903 Control address is specific to the linksys type and is used for sending the control
2904 packets unique to them.
2906 Security Events Framework
2907 -------------------------
2908 * Asterisk has a new C API for reporting security events. The module res_security_log
2909 sends these events to the "security" logger level. Currently, AMI is the only
2910 Asterisk component that reports security events. However, SIP support will be
2911 coming soon. For more information on the security events framework, see the
2912 "Asterisk Security Framework" section of the Asterisk wiki at
2913 https://wiki.asterisk.org/wiki/x/wgBQ
2914 * SIP support was added in Asterisk 10
2915 * This API now supports IPv6 addresses
2919 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2920 * A spandsp based fax backend (res_fax_spandsp) has been added.
2921 * The app_fax module has been deprecated in favor of the res_fax module and
2922 the new res_fax_spandsp backend.
2923 * The SendFAX and ReceiveFAX applications now send their log messages to a
2924 'fax' logger level, instead of to the generic logger levels. To see these
2925 messages, the system's logger.conf file will need to direct the 'fax' logger
2926 level to one or more destinations; the logger.conf.sample file includes an
2927 example of how to do this. Note that if the 'fax' logger level is *not*
2928 directed to at least one destination, log messages generated by these
2929 applications will be lost, and that if the 'fax' logger level is directed to
2930 the console, the 'core set verbose' and 'core set debug' CLI commands will
2931 have no effect on whether the messages appear on the console or not.
2935 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2936 Now, in order to enable transmitting silence during record the transmit_silence
2937 option should be used. transmit_silence_during_record remains a valid option, but
2938 defaults to the behavior of the transmit_silence option.
2939 * Addition of the Unit Test Framework API for managing registration and execution
2940 of unit tests with the purpose of verifying the operation of C functions.
2941 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2942 XMPP text messages to the remote JID.
2943 * Modules.conf has a new option - "require" - that marks a module as critical for
2944 the execution of Asterisk.
2945 If one of the required modules fail to load, Asterisk will exit with a return
2947 * An 'X' option has been added to the asterisk application which enables #exec support.
2948 This allows #exec to be used in asterisk.conf.
2949 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2950 * A new lockconfdir option has been added to asterisk.conf to protect the
2951 configuration directory (/etc/asterisk by default) during reloads.
2952 * The parkeddynamic option has been added to features.conf to enable the creation
2953 of dynamic parkinglots.
2954 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2955 the reportalarms config option.
2956 * chan_dahdi supports dialing configuring and dialing by device file name.
2957 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2958 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2959 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2960 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2961 Handy for the above name-based syntax as it does not depend on
2962 initialization order.
2963 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2964 significant increase in performance (about 3X) for installations using this switchtype.
2965 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2966 AIS. For more information, please see the Distributed Device State section of the
2967 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2968 * The addition of G.719 pass-through support.
2969 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2970 during device configuration.
2971 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2972 have less than 3 lines on the LCD.
2973 * Realtime now supports database failover. See the sample extconfig.conf for details.
2974 * The addition of improved translation path building for wideband codecs. Sample
2975 rate changes during translation are now avoided unless absolutely necessary.
2976 * The addition of the res_stun_monitor module for monitoring and reacting to network
2977 changes while behind a NAT.
2978 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2979 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2980 These allow support for any Administration. Default is AT&T values.
2984 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2985 optionally accept a filename, to apply the setting only to the code generated from
2986 that source file when Asterisk was built. However, there are some modules in Asterisk
2987 that are composed of multiple source files, so this did not result in the behavior
2988 that users expected. In this version, 'core set debug' and 'core set verbose'
2989 can optionally accept *module* names instead (with or without the .so extension),
2990 which applies the setting to the entire module specified, regardless of which source
2991 files it was built from.
2992 * New 'manager show settings' command showing the current settings loaded from
2994 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2995 the channel hangup request to all channels.
2996 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2998 ------------------------------------------------------------------------------
2999 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3000 ------------------------------------------------------------------------------
3004 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3005 Snom phones use this for call pickup of extensions that the phone is
3007 * Added support for setting the domain in the URI for caller of an
3008 outbound call by using the SIPFROMDOMAIN channel variable.
3009 * Added a new configuration option "remotesecret" for authentication to
3010 remote services. For backwards compatibility, "secret" still has the
3011 same function as before, but now you can configure both a remote secret and a
3012 local secret for mutual authentication.
3013 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3014 the sound will be played to the target of an attended transfer
3015 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3016 finer control over how many peers Asterisk will qualify and the gap between them
3017 when all peers need to be qualified at the same time.
3018 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3019 (either globally or for a specific peer), chan_sip will treat any SDP data
3020 it receives as new data and update the media stream accordingly. By
3021 default, Asterisk will only modify the media stream if the SDP session
3022 version received is different from the current SDP session version. This
3023 option is required to interoperate with devices that have non-standard SDP
3024 session version implementations (observed with Microsoft OCS). This option
3025 is disabled by default.
3026 * The parsing of register => lines in sip.conf has been modified to allow a port
3027 to be present in the "user" portion. Please see the sip.conf.sample file for more
3029 * Added support for subscribing to MWI on a remote server and making the status available
3030 as a mailbox. Please see the sip.conf.sample file for more information.
3031 * Added a function to remove SIP headers added in the dialplan before the
3032 first INVITE is generated - SIPRemoveHeader()
3033 * Channel variables set with setvar= in a device configuration is now
3034 set both for inbound and outbound calls.
3035 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3039 * Added immediate option to iax.conf
3040 * Added forceencryption option to iax.conf
3041 * Added Encryption and Trunk status to manager command "iaxpeers"
3045 * The configuration file now holds separate sections for devices and lines.
3046 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3051 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3052 support for LibOpenR2. http://www.libopenr2.org/
3053 * The UK option waitfordialtone has been added for use with BT analog
3055 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3056 is used in conjunction with the 'faxdetect' configuration option. When
3057 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3058 switch to the configured faxbuffers policy. For example, to use 6 buffers
3059 and a 'full' buffer policy for a fax transmission, add:
3061 The faxbuffers configuration will be in affect until the call is torn down.
3062 * Added service message support for 4ESS/5ESS switches.
3066 * For DAHDI channels, the CHANNEL() dialplan function now
3067 supports changing the channel's buffer policy (for the current
3068 call only), using this syntax:
3070 exten => s,n,Set(CHANNEL(buffers)=6,full)
3072 This would change the channel to the 'full' buffer policy and
3073 6 (six) buffers. Possible options for this setting are the same
3074 as those in chan_dahdi.conf.
3075 * Added a new dialplan function, CURLOPT, which permits setting various
3076 options that may be useful with the CURL dialplan function, such as
3077 cookies, proxies, connection timeouts, passwords, etc.
3078 * Permit the syntax and synopsis fields of the corresponding dialplan
3079 functions to be individually set from func_odbc.conf.
3080 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3081 * func_odbc now may specify an insert query to execute, when the write query
3082 affects 0 rows (usually indicating that no such row exists).
3083 * Added a new dialplan function, LISTFILTER, which permits removing elements
3084 from a set list, by name. Uses the same general syntax as the existing CUT
3085 and FIELDQTY dialplan functions, which also manage lists.
3086 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3087 obtaining realtime data from the dialplan.
3088 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3089 a subroutine when using the GoSub() and Return() applications.
3090 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3091 of "core show function AUDIOHOOK_INHERIT" from the CLI
3092 * Added AES_ENCRYPT. For information on its use, please see the output
3093 of "core show function AES_ENCRYPT" from the CLI
3094 * Added AES_DECRYPT. For information on its use, please see the output
3095 of "core show function AES_DECRYPT" from the CLI
3096 * func_odbc now supports database transactions across multiple queries.
3100 * Scheduled meetme conferences may now have their end times extended by
3102 * app_authenticate now gives the ability to select a prompt other than
3104 * app_directory now pays attention to the searchcontexts setting in
3105 voicemail.conf and will look through all contexts, if no context is
3106 specified in the initial argument.
3107 * A new application, Originate, has been introduced, that allows asynchronous
3108 call origination from the dialplan.
3109 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3110 in addition to the setting in the "general" context.
3111 * Added ConfBridge dialplan application which does conference bridges without
3112 DAHDI. For information on its use, please see the output of
3113 "core show application ConfBridge" from the CLI.
3117 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3118 operation to the AMI Redirect action.
3119 * extensions.conf now allows you to use keyword "same" to define an extension
3120 without actually specifying an extension. It uses exactly the same pattern
3121 as previously used on the last "exten" line. For example:
3122 exten => 123,1,NoOp(something)
3123 same => n,SomethingElse()
3124 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3125 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3126 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3127 by the new clialiases module. See cli_aliases.conf.sample file.
3128 * Times within timespecs are now accurate down to the minute. This is a change
3129 from historical Asterisk, which only provided timespecs rounded to the nearest
3130 even (read: evenly divisible by 2) minute mark.
3131 * The realtime switch now supports an option flag, 'p', which disables searches for
3133 * In addition to a time range and date range, timespecs now accept a 5th optional
3134 argument, timezone. This allows you to perform time checks on alternate
3135 timezones, especially if those daylight savings time ranges vary from your
3136 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3138 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3139 give you the correct output for an asterisk box behind nat. It will give you the
3140 externhost and localnet settings.
3141 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3142 can connect calls in passthrough mode, as well as record and play back files.
3143 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3144 using pickupsound and pickupfailsound in features.conf.
3145 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3146 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3147 instead of the /var/run/asterisk.pid where it used to be. This will make
3148 installs as non-root easier to manage.
3153 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3154 be written; they will no longer be explicitly written.
3156 Asterisk Manager Interface
3157 --------------------------
3158 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3159 a non-empty value) in your request. If you do this, any pending AMI events will
3160 *not* be included in the response to your request as they would normally, but
3161 will be left in the event queue for the next request you make to retrieve. For
3162 some applications, this will allow you to guarantee that you will only see
3163 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3164 To know whether the Asterisk server supports this header or not, your client can
3165 inspect the first response back from the server to see if it includes this header:
3167 Pragma: SuppressEvents
3169 If this is included, the server supports event suppression.
3171 * Added 4 new Actions to list skinny device(s) and line(s)
3177 LDAP Schema File Additions
3178 --------------------------
3179 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3180 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3182 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3183 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3184 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3185 * Removed redundant IPaddr (there's already IPAddress)
3186 - Gives more configuration Flags for SIP-Users available (tested)
3187 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3188 without extensibleObject (which really should be the last resort); gives
3189 also additional possibilities for LDAP-filter
3191 ------------------------------------------------------------------------------
3192 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3193 ------------------------------------------------------------------------------
3195 Device State Handling
3196 ---------------------
3197 * The event infrastructure in Asterisk got another big update to help support
3198 distributed events. It currently supports distributed device state and
3199 distributed Voicemail MWI (Message Waiting Indication). A new module has
3200 been merged, res_ais, which facilitates communicating events between servers.
3201 It uses the SAForum AIS (Service Availability Forum Application Interface
3202 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3203 a cluster of Asterisk servers, and to share events between them. For more
3204 information on setting this up, refer to the Distributed Device State section
3205 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3209 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3210 variables from an Asterisk configuration file.
3211 * The JACK_HOOK function now has a c() option to supply a custom client name.
3212 * Added two new dialplan functions from libspeex for audio gain control and
3213 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3214 rx directions of a channel from the dialplan.
3215 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3216 based on other parameters. The default is still to search based on the
3217 forwarding station ID. However, there are new options that allow you to search
3218 based on the message desk terminal ID, or the message desk number.
3219 * TIMEOUT() has been modified to be accurate down to the millisecond.
3220 * ENUM*() functions now include the following new options:
3221 - 'u' returns the full URI and does not strip off the URI-scheme.
3222 - 's' triggers ISN specific rewriting
3223 - 'i' looks for branches into an Infrastructure ENUM tree
3224 - 'd' for a direct DNS lookup without any flipping of digits.
3225 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3226 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3227 deviation of jitter, rtt, and loss for a call using chan_sip.
3229 DAHDI channel driver (chan_dahdi) Changes
3230 ----------------------------------------
3231 * Channels can now be configured using named sections in chan_dahdi.conf, just
3232 like other channel drivers, including the use of templates.
3233 * The default for pridialplan has changed from 'national' to 'unknown'.
3237 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3238 to something that matches the pattern a hint will be created using the contents
3239 and variables evaluated.
3240 * Dialplan matching has been extended to allow an extension to return to the
3241 PBX core to wait for more digits. This is done by using the new dialplan
3242 application called "Incomplete". This will permit a whole new level of
3243 extension control, by giving the administrator more control over early
3244 matches employing one of the short-circuit pattern match operators. Note
3245 that custom applications can trigger this same behavior by returning the
3246 special value AST_PBX_INCOMPLETE.
3250 * Directory now permits both first and last names to be matched at the same
3251 time. In addition, the number of digits to enter of the name can be set in
3252 the arguments to Directory; previously, you could enter only 3, regardless
3253 of how many names are in your company. For large companies, this should be
3255 * Voicemail now permits a mailbox setting to wrap around from first to last
3256 messages, if the "messagewrap" option is set to a true value.
3257 * Voicemail now permits an external script to be run, for password validation.
3258 The script should output "VALID" or "INVALID" on stdout, depending upon the
3259 wish to validate or invalidate the password given. Arguments are:
3260 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3262 * Dial has a new option: F(context^extension^pri), which permits a callee to
3263 continue in the dialplan, at the specified label, if the caller hangs up.
3264 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3265 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3266 * The Jack application now has a c() option to supply a custom client name.
3267 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3268 like the pre-existing whisper mode, except that the spy can also talk to the
3269 participant on the bridged channel as well.
3270 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3271 to be spoken instead of the channel name or number. For more information on the
3272 use of this option, issue the command "core show application ChanSpy" from the
3274 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3275 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3276 words, if using the 'd' option, it is not possible to enter a number to append to
3277 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3278 change to whisper mode, and pressing 6 will change to barge mode.
3279 * ExternalIVR now takes several options that affect the way it performs, as
3280 well as having several new commands. Please see the External IVR page on the Asterisk
3281 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3282 * Added ability to communicate over a TCP socket instead of forking a child process for the
3283 ExternalIVR application.
3284 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3285 of just the first one if you give the function more then one channel to check.
3286 * PrivacyManager now takes an option where you can specify a context where the
3287 given number will be matched. This way you have more control over who is allowed
3288 and it stops the people who blindly enter 10 digits.
3289 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3290 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3291 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3292 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3293 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3294 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3295 * The Dial() application no longer copies the language used by the caller to the callee's
3296 channel. If you desire for the caller's channel's language to be used for file playback
3297 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3298 * SendImage() no longer hangs up the channel on error; instead, it sets the
3299 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3300 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3302 * Park has a new option, 's', which silences the announcement of the parking space number.
3303 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3304 invalid input and will be assumed to mean that no timeout is desired.
3308 * Added DNS manager support to registrations for peers referencing peer entries.
3309 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3310 as well as periodically updating the IP address. These properties allow for
3311 better performance as well as recovery in the event of an IP change.
3312 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3313 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3314 These changes also provide performance improvements for call setup and tear down.
3315 * Added ability to specify registration expiry time on a per registration basis in
3317 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3319 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3320 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3321 * 'sip show peers' and 'sip show users' display their entries sorted in
3322 alphabetical order, as opposed to the order they were in, in the config
3324 * Videosupport now supports an additional option, "always", which always sets
3325 up video RTP ports, even on clients that don't support it. This helps with
3326 callfiles and certain transfers to ensure that if two video phones are
3327 connected, they will always share video feeds.
3331 * Existing DNS manager lookups extended to check for SRV records.
3332 * IAX2 encryption support has been improved to support periodic key rotation
3333 within a call for enhanced security. The option "keyrotate" has been
3334 provided to disable this functionality to preserve backwards compatibility
3335 with older versions of IAX2 that do not support key rotation.
3339 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3340 data tree based on the given <path>.
3341 * New CLI command "data show providers" that will display all the registered
3343 * New CLI command, "config reload <file.conf>" which reloads any module that
3344 references that particular configuration file. Also added "config list"
3345 which shows which configuration files are in use.
3346 * New CLI commands, "pri show version" and "ss7 show version" that will
3347 display which version of libpri and libss7 are being used, respectively.
3348 A new API call was added so trunk will now have to be compiled against
3349 a versions of libpri and libss7 that have them or it will not know that
3350 these libraries exist.
3351 * The commands "core show globals", "core set global" and "core set chanvar" has
3352 been deprecated in favor of the more semanticly correct "dialplan show globals",
3353 "dialplan set chanvar" and "dialplan set global".
3354 * New CLI command "dialplan show chanvar" to list all variables associated
3355 with a given channel.
3359 * Addresses managed by DNS manager now can check to see if there is a DNS
3360 SRV record for a given domain and will use that hostname/port if present.
3362 AMI - The manager (TCP/TLS/HTTP)
3363 --------------------------------
3364 * The Status command now takes an optional list of variables to display
3365 along with channel status.
3366 * The QueueEntry event now also includes the channel's uniqueid
3370 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3371 as some people were running into this limit. This limit has been increased
3376 * The TRANSFER queue log entry now includes the the caller's original
3377 position in the transferred-from queue.
3378 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3379 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3380 as well as an explanation about timeout options in general
3381 * Added a new option - C - for forcing the "answered elsewhere" flag on
3382 cancellation of calls in to members of the queue. This is to avoid the
3383 call to a member of a queue having the call listed as a "missed call".
3387 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3388 adaptive capabilities. What this means in practical terms is that if your
3389 realtime table lacks critical fields, Asterisk will now emit warnings to
3390 that effect. Also, some of the realtime drivers have the ability (if
3391 configured) to automatically add those columns to the table with the
3392 correct type and length.
3396 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3397 the 'setvar' option to cause a given audio file to be played upon completion
3398 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3399 Skinny channels only.
3400 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3401 Hoard page on the Asterisk wiki for more information:
3402 https://wiki.asterisk.org/wiki/x/pQBB
3403 * Config file variables may now be appended to, by using the '+=' append
3404 operator. This is most helpful when working with long SQL queries in
3405 func_odbc.conf, as the queries no longer need to be specified on a single
3407 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3408 which will add a second to the billsec when the ending
3409 time is set, if the number in the microseconds field of the end time is
3410 greater than the number of microseconds in the answer time. This allows
3411 users to count the 'initiated' seconds in their billing records.
3413 ------------------------------------------------------------------------------
3414 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3415 ------------------------------------------------------------------------------
3417 AMI - The manager (TCP/TLS/HTTP)
3418 --------------------------------
3419 * Manager has undergone a lot of changes, all of them documented
3420 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3421 * Manager version has changed to 1.1
3422 * Added a new action 'CoreShowChannels' to list currently defined channels
3423 and some information about them.
3424 * Added a new action 'SIPshowregistry' to list SIP registrations.
3425 * Added TLS support for the manager interface and HTTP server
3426 * Added the URI redirect option for the built-in HTTP server
3427 * The output of CallerID in Manager events is now more consistent.
3428 CallerIDNum is used for number and CallerIDName for name.
3429 * Enable https support for builtin web server.
3430 See configs/http.conf.sample for details.
3431 * Added a new action, GetConfigJSON, which can return the contents of an
3432 Asterisk configuration file in JSON format. This is intended to help
3433 improve the performance of AJAX applications using the manager interface
3435 * SIP and IAX manager events now use "ChannelType" in all cases where we
3436 indicate channel driver. Previously, we used a mixture of "Channel"
3437 and "ChannelDriver" headers.
3438 * Added a "Bridge" action which allows you to bridge any two channels that
3439 are currently active on the system.
3440 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3441 the voicemail users setup.
3442 * Added 'DBDel' and 'DBDelTree' manager commands.
3443 * cdr_manager now reports events via the "cdr" level, separating it from
3444 the very verbose "call" level.
3445 * Manager users are now stored in memory. If you change the manager account
3446 list (delete or add accounts) you need to reload manager.
3447 * Added Masquerade manager event for when a masquerade happens between
3449 * Added "manager reload" command for the CLI
3450 * Lots of commands that only provided information are now allowed under the
3451 Reporting privilege, instead of only under Call or System.
3452 * The IAX* commands now require either System or Reporting privilege, to
3453 mirror the privileges of the SIP* commands.
3454 * Added ability to retrieve list of categories in a config file.
3455 * Added ability to retrieve the content of a particular category.
3456 * Added ability to empty a context.
3457 * Created new action to create a new file.
3458 * Updated delete action to allow deletion by line number with respect to category.
3459 * Added new action insert to add new variable to category at specified line.
3460 * Updated action newcat to allow new category to be inserted in file above another
3462 * Added new event "JitterBufStats" in the IAX2 channel
3463 * Originate now requires the Originate privilege and, if you want to call out
3464 to a subshell, it requires the System privilege, as well. This was done to
3465 enhance manager security.
3466 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3467 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3468 or manager show command Atxfer from the CLI
3469 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3470 details or manager show command IAXregistry
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