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 ------------------------------------------------------------------------------
15 ------------------------------------------------------------------------------
16 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
17 ------------------------------------------------------------------------------
21 * A new Playback URI 'tone' has been added. Tones are specified either as
22 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
23 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
24 URIs in that they must be stopped manually and will continue to occupy
25 a channel's ARI control queue until they are stopped. They also can not
26 be rewound or fastforwarded.
28 ------------------------------------------------------------------------------
29 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
30 ------------------------------------------------------------------------------
33 --------------------------
34 * Record application now has an option 'o' which allows 0 to act as an exit
35 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
36 * Monitor() - A new option, B(), has been added that will turn on a periodic
37 beep while the call is being recorded.
40 --------------------------
41 * A new function was added: PERIODIC_HOOK. This allows running a periodic
42 dialplan hook on a channel. Any audio generated by this hook will be
43 injected into the call.
46 --------------------------
47 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
48 as the chanprefix parameter if the 'u' option is specified.
51 --------------------------
52 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
53 conference user menus.
55 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
56 menus, bridge settings, and user settings that have been applied by the
57 CONFBRIDGE dialplan function.
59 * The ConfBridge dialplan application now sets a channel variable,
60 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
61 how a channel exited the conference.
63 * Added conference user option 'announce_join_leave_review'. This option
64 implies 'announce_join_leave' with the added effect that the user will
65 be asked if they want to confirm or re-record the recording of their
66 name when entering the conference
69 --------------------------
70 * At exit, the Directory application now sets a channel variable
71 DIRECTORY_RESULT to one of the following based on the reason for exiting:
72 OPERATOR user requested operator by pressing '0' for operator
73 ASSISTANT user requested assistant by pressing '*' for assistant
74 TIMEOUT user pressed nothing and Directory stopped waiting
75 HANGUP user's channel hung up
76 SELECTED user selected a user from the directory and is routed
77 USEREXIT user pressed '#' from the selection prompt to exit
78 FAILED directory failed in a way that wasn't accounted for. Dang.
81 --------------------------
82 * MusicOnHold streams (all modes other than "files") now support wide band
86 --------------------------
87 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
88 and for the channel executing Page respectively.
91 --------------------------
92 * PickupChan now accepts channel uniqueids of channels to pickup.
95 --------------------------
96 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
97 to 'true' (case insensitive), then any Say application (SayNumber,
98 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
99 anticipate DTMF. If DTMF is received, these applications will behave like
100 the background application and jump to the received extension once a match
101 is established or after a short period of inactivity.
104 -------------------------
105 * A new function, MIXMONITOR, has been added to allow access to individual
106 instances of MixMonitor on a channel.
107 * A new option, B(), has been added that will turn on a periodic beep while the
108 call is being recorded.
112 -------------------------
115 -------------------------
116 * TEL URI support for inbound INVITE requests has been added. chan_sip will
117 now handle TEL schemes in the Request and From URIs. The phone-context in
118 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
122 -------------------------
123 * Core Show Locks output now includes Thread/LWP ID if the platform
124 supports this feature.
125 * New "logger add channel" and "logger remove channel" CLI commands have
126 been added to allow creation and deletion of dynamic logger channels
127 without configuration changes. These dynamic logger channels will only
128 exist until the next restart of asterisk.
132 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
133 the new AST_SORCERY diaplan function.
137 * The live recording object on recording events now contains a target_uri
138 field which contains the URI of what is being recorded.
140 * The bridge type used when creating a bridge is now a comma separated list of
141 bridge properties. Valid options are: mixing, holding, dtmf_events, and
144 * A channelId can now be provided when creating a channel, either in the
145 uri (POST channels/my-channel-id) or as query parameter. A local channel
146 will suffix the second channel id with ';2' unless provided as query
147 parameter otherChannelId.
149 * A bridgeId can now be provided when creating a bridge, either in the uri
150 (POST bridges/my-bridge-id) or as a query parameter.
152 * A playbackId can be provided when starting a playback, either in the uri
153 (POST channels/my-channel-id/play/my-playback-id) or as a query parameter.
155 * A snoop channel can be started with a snoopId, in the uri or query.
159 * Originate now takes optional parameters ChannelId and OtherChannelId,
160 used to set the UniqueId on creation. The other id is assigned to the
161 second channel when dialing LOCAL, or defaults to appending ;2 if only
162 the single Id is given.
164 * The Mixmonitor action now has a "Command" header that can be used to
165 indicate a post-process command to run once recording finishes.
169 * A new set of Alembic scripts has been added for CDR tables. This will create
170 a 'cdr' table with the default schema that Asterisk expects.
174 * A new module, res_hep, has been added, that acts as a generic packet
175 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
176 It can be configured via hep.conf. Other modules can use res_hep to send
177 message traffic to a HEP capture server.
181 * A new module, res_hep_pjsip, has been added that will forward PJSIP
182 message traffic to a HEP capture server. See res_hep for more
187 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
188 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
190 * Added the following new CLI commands:
191 - "pjsip show contacts" - list all current PJSIP contacts.
192 - "pjsip show contact" - show specific information about a current PJSIP
194 - "pjsip show channel" - show detailed information about a PJSIP channel.
198 * A new module, res_pjsip_multihomed handles situations where the system
199 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
200 determines which interface should be used during message sending.
202 res_pjsip_pidf_digium_body_supplement
204 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
205 request body formatting for presence support in Digium phones.
207 res_pjsip_send_to_voicemail
209 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
210 particular headers to transfer a PJSIP channel directly to a particular
211 extension that has VoiceMail. This is intended to be used with Digium
212 phones that support this feature.
214 res_pjsip_outbound_registration
216 * A new CLI command has been added: "pjsip show registrations", which lists
217 all configured PJSIP registrations
220 ------------------------------------------------------------------------------
221 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
222 ------------------------------------------------------------------------------
226 * Added a new module that provides AMI control over MWI within Asterisk,
227 res_mwi_external_ami. Note that this module depends on res_mwi_external;
228 for more information on enabling this module, see res_mwi_external.
229 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
230 the MWIGet/MWIGetComplete events.
232 * The DialStatus field in the DialEnd event can now contain additional
233 statuses that convey how the dial operation terminated. This includes
234 ABORT, CONTINUE, and GOTO.
236 * AMI will now emit security events. A new class authorization has been
237 added in manager.conf for the security events, 'security'. The new events
239 - FailedACL - raised when a request violates an ACL check
240 - InvalidAccountID - raised when a request fails an authentication
241 check due to an invalid account ID
242 - SessionLimit - raised when a request fails due to exceeding the
243 number of allowed concurrent sessions for a service
244 - MemoryLimit - raised when a request fails due to an internal memory
246 - LoadAverageLimit - raised when a request fails because a configured
247 load average limit has been reached
248 - RequestNotAllowed - raised when a request is not allowed by
250 - AuthMethodNotAllowed - raised when a request used an authentication
251 method not allowed by the service
252 - RequestBadFormat - raised when a request is received with bad formatting
253 - SuccessfulAuth - raised when a request successfully authenticates
254 - UnexpectedAddress - raised when a request has a different source address
255 then what is expected for a session already in progress with a service
256 - ChallengeResponseFailed - raised when a request's attempt to authenticate
257 has been challenged, and the request failed the authentication challenge
258 - InvalidPassword - raised when a request provides an invalid password
259 during an authentication attempt
260 - ChallengeSent - raised when an Asterisk service send an authentication
261 challenge to a request
262 - InvalidTransport - raised when a request attempts to use a transport not
263 allowed by the Asterisk service
265 * Bridge related events now have two additional fields: BridgeName and
266 BridgeCreator. BridgeName is a descriptive name for the bridge;
267 BridgeCreator is the name of the entity that created the bridge. This
268 affects the following events: ConfbridgeStart, ConfbridgeEnd,
269 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
270 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
271 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
275 * The Bridge data model now contains the additional fields 'name' and
276 'creator'. The 'name' field conveys a descriptive name for the bridge;
277 the 'creator' field conveys the name of the entity that created the bridge.
278 This affects all responses to HTTP requests that return a Bridge data model
279 as well as all event derived data models that contain a Bridge data model.
280 The POST /bridges operation may now optionally specify a name to give to
281 the bridge being created.
283 * Added a new ARI resource 'mailboxes' which allows the creation and
284 modification of mailboxes managed by external MWI. Modules res_mwi_external
285 and res_stasis_mailbox must be enabled to use this resource. For more
286 information on external MWI control, see res_mwi_external.
288 * Added new events for externally initiated transfers. The event
289 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
290 of a bridge in the ARI controlled application to the dialplan; the
291 BridgeAttendedTransfer event is raised when a channel initiates an
292 attended transfer of a bridge in the ARI controlled application to the
295 * Channel variables may now be specified as a body parameter to the
296 POST /channels operation. The 'variables' key in the JSON is interpreted
297 as a sequence of key/value pairs that will be added to the created channel
298 as channel variables. Other parameters in the JSON body are treated as
299 query parameters of the same name.
303 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
304 automatically handled by the HTTP server if a request is received with a
305 Transfer-Encoding type of "chunked".
309 * Path support has been added with the 'support_path' option in registration
312 * A 'debug' option has been added to the globals section that will allow
313 sip messages to be logged.
315 * A 'set_var' option has been added to endpoints that will automatically
316 set the desired variable(s) on a channel created for that endpoint.
318 * Several new tables and columns have been added to the realtime schema for
319 the res_pjsip related modules. See the UPGRADE.txt notes for updating
324 * A new module, res_mwi_external, has been added to Asterisk. This module
325 acts as a base framework that other modules can build on top of to allow
326 an external system to control MWI within Asterisk. For implementations
327 that make use of res_mwi_external, see res_mwi_external_ami and
328 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
329 that may produce MWI themselves, such as app_voicemail. res_mwi_external
330 and other modules that depend on it cannot be built or loaded with
331 app_voicemail present.
335 * DNS functionality will now automatically be enabled if the system configured
336 nameservers can be retrieved. If the system configured nameservers can not be
337 retrieved the functionality will resort to using system resolution. Functionalty
338 such as SRV records and failover will not be available if system resolution
341 ------------------------------------------------------------------------------
342 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
343 ------------------------------------------------------------------------------
348 Asterisk 12 is a standard release of the Asterisk project. As such, the
349 focus of development for this release was on core architectural changes and
350 major new features. This includes:
351 * A more flexible bridging core based on the Bridging API
352 * A new internal message bus, Stasis
353 * Major standardization and consistency improvements to AMI
354 * Addition of the Asterisk RESTful Interface (ARI)
355 * A new SIP channel driver, chan_pjsip
356 In addition, as the vast majority of bridging in Asterisk was migrated to the
357 Bridging API used by ConfBridge, major changes were made to most of the
358 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
360 Specifications have been written for the affected interfaces. These
361 specifications are available on the Asterisk wiki:
362 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
363 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
364 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
366 It is *highly* recommended that anyone migrating to Asterisk 12 read the
367 information regarding its release both in this file and in the accompanying
368 UPGRADE.txt file. More detailed information on the major changes can be found
369 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
374 * Added build option DISABLE_INLINE. This option can be used to work around a
375 bug in gcc. For more information, see
376 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
378 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
379 the CHANNEL_TRACE build option were incompatible with the new bridging
382 * Asterisk now optionally uses libxslt to improve XML documentation generation
383 and maintainability. If libxslt is not available on the system, some XML
384 documentation will be incomplete.
386 * Asterisk now depends on libjansson. If a package of libjansson is not
387 available on your distro, please see http://www.digip.org/jansson/.
389 * Asterisk now depends on libuuid and, optionally, uriparser. It is
390 recommended that you install uriparser, even if it is optional.
392 * The new SIP stack and channel driver uses a particular version of PJSIP.
393 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
394 configuring and installing PJSIP for usage with Asterisk.
396 * Optional API was re-implemented to be more portable, and no longer requires
397 weak reference support from the compiler. The build option OPTIONAL_API may
398 be disabled to disable Optional API support.
405 * Along with AgentRequest, this application has been modified to be a
406 replacement for chan_agent. The act of a channel calling the AgentLogin
407 application places the channel into a pool of agents that can be
408 requested by the AgentRequest application. Note that this application, as
409 well as all other agent related functionality, is now provided by the
410 app_agent_pool module. See chan_agent and AgentRequest for more information.
412 * This application no longer performs agent authentication. If authentication
413 is desired, the dialplan needs to perform this function using the
414 Authenticate or VMAuthenticate application or through an AGI script before
417 * If this application is called and the agent is already logged in, the
418 dialplan will continue exection with the AGENT_STATUS channel variable set
419 to ALREADY_LOGGED_IN.
421 * The agents.conf schema has changed. Rather than specifying agents on a
422 single line in comma delineated fashion, each agent is defined in a separate
423 context. This allows agents to use the power of context templates in their
426 * A number of parameters from agents.conf have been removed. This includes
427 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
428 urlprefix, and savecallsin. These options were obsoleted by the move from
429 a channel driver model to the bridging/application model provided by
434 * A new application, this will request a logged in agent from the pool and
435 bridge the requested channel with the channel calling this application.
436 Logged in agents are those channels that called the AgentLogin application.
437 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
438 application will be set with an appropriate error value.
442 * This application has been removed. It was a holdover from when
443 AgentCallbackLogin was removed.
447 * Added support for additional Ademco DTMF signalling formats, including
448 Express 4+1, Express 4+2, High Speed and Super Fast.
450 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
451 call time, in milliseconds, to run the application.
453 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
454 maximum number of times to retry the call.
456 * Added a new configuration option answait. If set, the AlarmReceiver
457 application will wait the number of milliseconds specified by answait
458 after the channel has answered. Valid values range between 500
459 milliseconds and 10000 milliseconds.
461 * Added configuration option no_group_meta. If enabled, grouping of metadata
462 information in the AlarmReceiver log file will be skipped.
466 * It is now no longer possible to bypass updating the CDR on the channel
467 when answering. CDRs reflect the state of the channel and will always
468 reflect the time they were Answered.
472 * A new application in Asterisk, this will place the calling channel
473 into a holding bridge, optionally entertaining them with some form of
474 media. Channels participating in a holding bridge do not interact with
475 other channels in the same holding bridge. Optionally, however, a channel
476 may join as an announcer. Any media passed from an announcer channel is
477 played to all channels in the holding bridge. Channels leave a holding
478 bridge either when an optional timer expires, or via the ChannelRedirect
479 application or AMI Redirect action.
483 * All participants in a bridge can now be kicked out of a conference room
484 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
485 command, i.e., 'confbridge kick <conference> all'
487 * CLI output for the 'confbridge list' command has been improved. When
488 displaying information about a particular bridge, flags will now be shown
489 for the participating users indicating properties of that user.
491 * The ConfbridgeList event now contains the following fields: WaitMarked,
492 EndMarked, and Waiting. This displays additional properties about the
493 user's profile, as well as whether or not the user is waiting for a
494 Marked user to enter the conference.
496 * Added a new option for conference recording, record_file_append. If enabled,
497 when the recording is stopped and then re-started, the existing recording
498 will be used and appended to.
500 * ConfBridge now has the ability to set the language of announcements to the
501 conference. The language can be set on a bridge profile in confbridge.conf
502 or by the dialplan function CONFBRIDGE(bridge,language)=en.
506 * The channel variable CPLAYBACKSTATUS may now return the value
507 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
508 such as AMI. See the AMI action ControlPlayback for more information.
512 * Added the 'a' option, which allows the caller to enter in an additional
513 alias for the user in the directory. This option must be used in conjunction
514 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
515 specified in voicemail.conf.
519 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
520 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
521 containing the unique ID of the bridge that the channel happens to be in.
525 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
526 for more information.
528 * Variables are no longer purged from the original CDR. See the 'v' option for
531 * The 'A' option has been removed. The Answer time on a CDR is never updated
534 * The 'd' option has been removed. The disposition on a CDR is a function of
535 the state of the channel and cannot be altered.
537 * The 'D' option has been removed. Who the Party B is on a CDR is a function
538 of the state of the respective channels involved in the CDR and cannot be
541 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
542 such that the start time and, if applicable, the answer time was updated.
543 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
544 'r' option now triggers the Reset, setting the start time (and answer time
545 if applicable) to the current time. Note that the 'a' option still sets
546 the answer time to the current time if the channel was already answered.
548 * The 's' option has been removed. A variable can be set on the original CDR
549 if desired using the CDR function, and removed from a forked CDR using the
552 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
553 longer applies in the CDR engine.
555 * The 'v' option now prevents the copy of the variables from the original CDR
556 to the forked CDR. Previously the variables were always copied but were
557 removed from the original. This was changed as removing variables from a CDR
558 can have unintended side effects - this option allows the user to prevent
559 propagation of variables from the original to the forked without modifying
564 * Added the 'n' option to MeetMe to prevent application of the DENOISE
565 function to a channel joining a conference. Some channel drivers that vary
566 the number of audio samples in a voice frame will experience significant
567 quality problems if a denoiser is attached to the channel; this option gives
568 them the ability to remove the denoiser without having to unload func_speex.
572 * The 'b' option now includes conferences as well as sounds played to the
575 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
576 running during a transfer. If a MixMonitor is started on a channel,
577 the MixMonitor will continue to record the audio passing through the
578 channel even in the presence of transfers.
582 * The NoCDR application is deprecated. Please use the CDR_PROP function to
585 * While the NoCDR application will prevent CDRs for a channel from being
586 propagated to registered CDR backends, it will not prevent that data from
587 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
588 function that enables CDRs on a channel will restore those records that have
589 not yet been finalized.
593 * The app_parkandannounce module has been removed. The application
594 ParkAndAnnounce is now provided by the res_parking module. See the
595 res_parking changes for more information.
599 * Added queue available hint. The hint can be added to the dialplan using the
600 following syntax: exten,hint,Queue:{queue_name}_avail
601 For example, if the name of the queue is 'markq':
602 exten => 8501,hint,Queue:markq_avail
603 This will report 'InUse' if there are no logged in agents or no free agents.
604 It will report 'Idle' when an agent is free.
606 * Queues now support a hint for member paused state. The hint uses the form
607 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
608 are the name of the queue and the name of the member to subscribe to,
609 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
610 Members will show as In Use when paused.
612 * The configuration options eventwhencalled and eventmemberstatus have been
613 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
614 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
615 sent. The "Variable" fields will also no longer exist on the Agent* events.
616 These events can be filtered out from a connected AMI client using the
617 eventfilter setting in manager.conf.
619 * The queue log now differentiates between blind and attended transfers. A
620 blind transfer will result in a BLINDTRANSFER message with the destination
621 context and extension. An attended transfer will result in an
622 ATTENDEDTRANSFER message. This message will indicate the method by which
623 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
624 for running an application on a bridge or channel, or "LINK" for linking
625 two bridges together with local channels. The queue log will also now detect
626 externally initiated blind and attended transfers and record the transfer
629 * When performing queue pause/unpause on an interface without specifying an
630 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
631 least one member of any queue exists for that interface.
633 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
634 for realtime queue log entries.
638 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
639 CDRs when they were previously disabled on a channel.
641 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
642 backends occurs on an as-needed basis in order to preserve linkedid
643 propagation and other needed behavior.
647 * A new application, this is similar to SayAlpha except that it supports
648 case sensitive playback of the specified characters. For example,
649 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
653 * This application is deprecated in favor of CHANNEL(amaflags).
657 * The SendDTMF application will now accept 'W' as valid input. This will cause
658 the application to delay one second while streaming DTMF.
662 * A new application in Asterisk 12, this hands control of the channel calling
663 the application over to an external system. Currently, external systems
664 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
668 * UserEvent will now handle duplicate keys by overwriting the previous value
671 * In addition to AMI, UserEvent invocations will now be distributed to any
672 interested Stasis applications.
676 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
677 system as mailbox@context. The rest of the system cannot add @default
678 to mailbox identifiers for app_voicemail that do not specify a context
679 any longer. It is a mailbox identifier format that should only be
680 interpreted by app_voicemail.
682 * The voicemail.conf configuration file now has an 'alias' configuration
683 parameter for use with the Directory application. The voicemail realtime
684 database table schema has also been updated with an 'alias' column.
689 * Pass through support has been added for both VP8 and Opus.
691 * Added format attribute negotiation for the Opus codec. Format attribute
692 negotiation is provided by the res_format_attr_opus module.
697 * Masquerades as an operation inside Asterisk have been effectively hidden
698 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
699 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
700 dropping of frame/audio hooks, and other internal implementation details
701 that users had to deal with. This fundamental change has large implications
702 throughout the changes documented for this version. For more information
703 about the new core architecture of Asterisk, please see the Asterisk wiki.
705 * Multiple parties in a bridge may now be transferred. If a participant in a
706 multi-party bridge initiates a blind transfer, a Local channel will be used
707 to execute the dialplan location that the transferer sent the parties to. If
708 a participant in a multi-party bridge initiates an attended transfer,
709 several options are possible. If the attended transfer results in a transfer
710 to an application, a Local channel is used. If the attended transfer results
711 in a transfer to another channel, the resulting channels will be merged into
714 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
715 driver specific. If the channel variable is set on the transferrer channel,
716 the sound will be played to the target of an attended transfer.
718 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
719 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
720 listed. Any more peers in the bridge will not be included in the list.
721 BRIDGEPEER is not valid in holding bridges like parking since those channels
722 do not talk to each other even though they are in a bridge.
724 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
725 and will contain a value if the BRIDGEPEER's channel driver supports it.
727 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
728 was responsible for an attended transfer in a similar fashion to
731 * Modules using the Configuration Framework or Sorcery must have XML
732 configuration documentation. This configuration documentation is included
733 with the rest of Asterisk's XML documentation, and is accessible via CLI
734 commands. See the CLI changes for more information.
736 AMI (Asterisk Manager Interface)
738 * Major changes were made to both the syntax as well as the semantics of the
739 AMI protocol. In particular, AMI events have been substantially improved
740 in this version of Asterisk. For more information, please see the AMI
741 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
743 * AMI events that reference a particular channel or bridge will now always
744 contain a standard set of fields. When multiple channels or bridges are
745 referenced in an event, fields for at least some subset of the channels
746 and bridges in the event will be prefixed with a descriptive name to avoid
747 name collisions. See the AMI event documentation on the Asterisk wiki for
750 * The CLI command 'manager show commands' no longer truncates command names
751 longer than 15 characters and no longer shows authorization requirement
752 for commands. 'manager show command' now displays the privileges needed
753 for using a given manager command instead.
755 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
756 peer in its response if the peer has a subscribe context set.
758 * The SIPqualifypeer action now acknowledges the request once it has
759 established that the request is against a known peer. It also issues a new
760 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
762 * The PlayDTMF action now supports an optional 'Duration' parameter. This
763 specifies the duration of the digit to be played, in milliseconds.
765 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
766 updates when changes occur instead of requiring the use of pollmailboxes.
768 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
769 AMI client to manipulate audio currently being played back on a channel. The
770 supported operations depend on the application being used to send audio to
771 the channel. When the audio playback was initiated using the ControlPlayback
772 application or CONTROL STREAM FILE AGI command, the audio can be paused,
773 stopped, restarted, reversed, or skipped forward. When initiated by other
774 mechanisms (such as the Playback application), the audio can be stopped,
775 reversed, or skipped forward.
777 * Channel related events now contain a snapshot of channel state, adding new
778 fields to many of these events.
780 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
781 in a future release. Please use the common 'Exten' field instead.
783 * The AMI event 'UserEvent' from app_userevent now contains the channel state
784 fields. The channel state fields will come before the body fields.
786 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
787 'UnParkedCall' have changed significantly in the new res_parking module.
789 The 'Channel' and 'From' headers are gone. For the channel that was parked
790 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
791 has a number of fields associated with it. The old 'Channel' header relayed
792 the same data as the new 'ParkeeChannel' header.
794 The 'From' field was ambiguous and changed meaning depending on the event.
795 for most of these, it was the name of the channel that parked the call
796 (the 'Parker'). There is no longer a header that provides this channel name,
797 however the 'ParkerDialString' will contain a dialstring to redial the
798 device that parked the call.
800 On UnParkedCall events, the 'From' header would instead represent the
801 channel responsible for retrieving the parkee. It receives a channel
802 snapshot labeled 'Retriever'. The 'from' field is is replaced with
805 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
807 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
808 fashion has changed the field names 'StartExten' and 'StopExten' to
809 'StartSpace' and 'StopSpace' respectively.
811 * The deprecated use of | (pipe) as a separator in the channelvars setting in
812 manager.conf has been removed.
814 * Channel Variables conveyed with a channel no longer contain the name of the
815 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
816 ChanVariable: bar=baz. When multiple channels are present in a single AMI
817 event, the various ChanVariable fields will contain a suffix that specifies
818 which channel they correspond to.
820 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
821 event always conveys the AMI event for a particular channel.
823 * All 'Reload' events have been consolidated into a single event type. This
824 event will always contain a Module field specifying the name of the module
825 and a Status field denoting the result of the reload. All modules now issue
826 this event when being reloaded.
828 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
829 fail to receive this event due to being connected after modules have loaded.
830 AMI connections that want to know when Asterisk is ready should listen for
831 the 'FullyBooted' event.
833 * app_fax now sends the same send fax/receive fax events as res_fax. The
834 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
835 now the 'ReceiveFAX' event.
837 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
838 'MusicOnHoldStop'. The sub type field has been removed.
840 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
841 carrier for another protocol.
843 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
844 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
845 to the specific channel. 'Both' may be specified to play a tone to both
846 channels. The old 'yes' option is still accepted as a way of playing the
847 tone to Channel2 only.
849 * The AMI 'Status' response event to the AMI Status action replaces the
850 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
851 indicate what bridge the channel is currently in.
853 * The AMI 'Hold' event has been moved out of individual channel drivers, into
854 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
857 * The AMI events in app_queue have been made more consistent with each other.
858 Events that reference channels (QueueCaller* and Agent*) will show
859 information about each channel. The (infamous) 'Join' and 'Leave' AMI
860 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
862 * The 'MCID' AMI event now publishes a channel snapshot when available and
863 its non-channel-snapshot parameters now use either the "MCallerID" or
864 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
865 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
866 parameters in the channel snapshot.
868 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
869 'AgentLogin' and 'AgentLogoff' respectively.
871 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
872 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
874 * 'ChannelUpdate' events have been removed.
876 * All AMI events now contain a 'SystemName' field, if available.
878 * Local channel optimization is now conveyed in two events:
879 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
880 when the Local channel driver begins attempting to optimize itself out of
881 the media path; the End event is sent after the channel halves have
882 successfully optimized themselves out of the media path.
884 * Local channel information in events is now prefixed with 'LocalOne' and
885 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
886 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
887 and 'LocalOptimizationEnd' events.
889 * The option 'allowmultiplelogin' can now be set or overriden in a particular
890 account. When set in the general context, it will act as the default
891 setting for defined accounts.
893 * The 'BridgeAction' event was removed. It technically added no value, as the
894 Bridge Action already receives confirmation of the bridge through a
895 successful completion Event.
897 * The 'BridgeExec' events were removed. These events duplicated the events that
898 occur in the Briding API, and are conveyed now through BridgeCreate,
899 BridgeEnter, and BridgeLeave events.
901 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
902 previous versions. They now report all SR/RR packets sent/received, and
903 have been restructured to better reflect the data sent in a SR/RR. In
904 particular, the event structure now supports multiple report blocks.
906 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
907 raised when a blind transfer/attended transfer completes successfully.
908 They contain information about the transfer that just completed, including
909 the location of the transfered channel.
911 * Added a 'security' class to AMI which outputs the required fields for
912 security messages similar to the log messages from res_security_log
914 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
915 that describes the status value in a human readable string.
917 CDR (Call Detail Records)
919 * Significant changes have been made to the behavior of CDRs. The CDR engine
920 was effectively rewritten and built on the Stasis message bus. For a full
921 definition of CDR behavior in Asterisk 12, please read the specification
922 on the Asterisk wiki (wiki.asterisk.org).
924 * CDRs will now be created between all participants in a bridge. For each
925 pair of channels in a bridge, a CDR is created to represent the path of
926 communication between those two endpoints. This lets an end user choose who
927 to bill for what during bridge operations with multiple parties.
929 * The duration, billsec, start, answer, and end times now reflect the times
930 associated with the current CDR for the channel, as opposed to a cumulative
931 measurement of all CDRs for that channel.
933 * When a CDR is dispatched, user defined CDR variables from both parties are
934 included in the resulting CDR. If both parties have the same variable, only
935 the Party A value is provided.
937 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
938 information regarding the CDR engine is logged as verbose messages. This
939 option should only be used if the behavior of the CDR engine needs to be
942 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
943 normally configured in cdr.conf.
945 * Added CLI command 'cdr show active {channel}'. When {channel} is not
946 specified, this command provides a summary of the channels with CDR
947 information and their statistics. When {channel} is specified, it shows
948 detailed information about all records associated with {channel}.
950 CEL (Channel Event Logging)
952 * CEL has undergone significant rework in Asterisk 12, and is now built on the
953 Stasis message bus. Please see the specification for CEL on the Asterisk
954 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
957 * The 'extra' field of all CEL events that use it now consists of a JSON blob
958 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
960 * BLINDTRANSFER events now report the transferee bridge unique
961 identifier, extension, and context in a JSON blob as the extra string
962 instead of the transferee channel name as the peer.
964 * ATTENDEDTRANSFER events now report the peer as NULL and additional
965 information in the 'extra' string as a JSON blob. For transfers that occur
966 between two bridged channels, the 'extra' JSON blob contains the primary
967 bridge unique identifier, the secondary channel name, and the secondary
968 bridge unique identifier. For transfers that occur between a bridged channel
969 and a channel running an app, the 'extra' JSON blob contains the primary
970 bridge unique identifier, the secondary channel name, and the app name.
972 * LOCAL_OPTIMIZE events have been added to convey local channel
973 optimizations with the record occurring for the semi-one channel and
974 the semi-two channel name in the peer field.
976 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
977 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
978 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
979 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
980 regardless of whether or not that bridge happens to contain multiple
985 * When compiled with '--enable-dev-mode', the astobj2 library will now add
986 several CLI commands that allow for inspection of ao2 containers that
987 register themselves with astobj2. The CLI commands are 'astobj2 container
988 dump', 'astobj2 container stats', and 'astobj2 container check'.
990 * Added specific CLI commands for bridge inspection. This includes 'bridge
991 show all', which lists all bridges in the system, and 'bridge show {id}',
992 which provides specific information about a bridge.
994 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
995 ejecting the channels currently in the bridge. If the channels cannot
996 continue in the dialplan or application that put them in the bridge, they
999 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1001 * Added commands to inspect and manipulate the registered bridge technologies.
1002 This include 'bridge technology show', which lists the registered bridge
1003 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1004 which controls whether or not a registered bridge technology can be used
1005 during smart bridge operations. If a technology is suspended, it will not
1006 be used when a bridge technology is picked for channels; when unsuspended,
1007 it can be used again.
1009 * The command 'config show help {module} {type} {option}' will show
1010 configuration documentation for modules with XML configuration
1011 documentation. When {module}, {type}, and {option} are omitted, a listing
1012 of all modules with registered documentation is displayed. When {module}
1013 is specified, a listing of all configuration types for that module is
1014 displayed, along with their synopsis. When {module} and {type} are
1015 specified, a listing of all configuration options for that type are
1016 displayed along with their synopsis. When {module}, {type}, and {option}
1017 are specified, detailed information for that configuration option is
1020 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1021 a listing of all installed media sounds available on the system and
1022 detailed information about a sound, respectively.
1024 * 'xmldoc dump' has been added. This CLI command will dump the XML
1025 documentation DOM as a string to the specified file. The Asterisk core
1026 will populate certain XML elements pulled from the source files with
1027 additional run-time information; this command lets a user produce the
1028 XML documentation with all information.
1032 * Parking has been pulled from core and placed into a separate module called
1033 res_parking. See Parking changes below for more details. Configuration for
1034 parking should now be performed in res_parking.conf. Configuration for
1035 parking in features.conf is now unsupported.
1037 * Core attended transfers now have several new options. While performing an
1038 attended transfer, the transferer now has the following options:
1039 - *1 - cancel the attended transfer (configurable via atxferabort)
1040 - *2 - complete the attended transfer, dropping out of the call
1041 (configurable via atxfercomplete)
1042 - *3 - complete the attended transfer, but stay in the call. This will turn
1043 the call into a multi-party bridge (configurable via atxferthreeway)
1044 - *4 - swap to the other party. Once an attended transfer has begun, this
1045 options may be used multiple times (configurable via atxferswap)
1047 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1048 must be on the channel initiating the transfer to have any effect.
1050 * The BRIDGE_FEATURES channel variable would previously only set features for
1051 the calling party and would set this feature regardless of whether the
1052 feature was in caps or in lowercase. Use of a caps feature for a letter
1053 will now apply the feature to the calling party while use of a lowercase
1054 letter will apply that feature to the called party.
1056 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1058 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1059 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1060 activated the dynamic feature.
1062 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1063 only on the channel executing the dynamic feature. Executing a dynamic
1064 feature on the bridge peer in a multi-party bridge will execute it on all
1065 peers of the activating channel.
1067 * You can now have the settings for a channel updated using the FEATURE()
1068 and FEATUREMAP() functions inherited to child channels by setting
1069 FEATURE(inherit)=yes.
1071 * automixmon now supports additional channel variables from automon including:
1072 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1073 and TOUCH_MIXMONITOR_MESSAGE_STOP
1075 * A new general features.conf option 'recordingfailsound' has been added which
1076 allowssetting a failure sound for a user tries to invoke a recording feature
1077 such as automon or automixmon and it fails.
1079 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1080 features.c for atxferdropcall=no to work properly. This option now just
1085 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1086 be used. Given that this can cause the Asterisk log files to grow quickly,
1087 this option should only be used if an external mechanism for log management
1092 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1093 will store the path information for that peer when it registers. Realtime
1094 tables can also use the 'supportpath' field to enable Path header support.
1096 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1097 objectIdentifier. This maps to the supportpath option in sip.conf.
1101 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1102 provides modules a useful abstraction on top of the many storage mechanisms
1103 in Asterisk, including the Asterisk Database, static configuration files,
1104 static Realtime, and dynamic Realtime. It also provides a caching service.
1105 Users can configure a hierarchy of data storage layers for specific modules
1108 * All future modules which utilize Sorcery for object persistence must have a
1109 column named "id" within their schema when using the Sorcery realtime module.
1110 This column must be able to contain a string of up to 128 characters in length.
1112 Security Events Framework
1114 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1115 the "seconds-microseconds" format that it was using previously.
1119 * The Stasis message bus is a publish/subscribe message bus internal to
1120 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1121 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1122 Stasis can be configured in stasis.conf. Note that these parameters operate
1123 at a very low level in Asterisk, and generally will not require changes.
1127 * When a channel driver is configured to enable jiterbuffers, they are now
1128 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1129 is already set for that channel when it enters, such as by the JITTERBUFFER
1130 function, then the existing jitterbuffer will be used and the one set by
1131 the channel driver will not be applied.
1135 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1136 dialplan applications provided by the app_agent_pool module. Agents are
1137 connected with callers using the new AgentRequest dialplan application.
1138 The Agents:<agent-id> device state is available to monitor the status of an
1139 agent. See agents.conf.sample for valid configuration options.
1141 * The updatecdr option has been removed. Altering the names of channels on a
1142 CDR is not supported - the name of the channel is the name of the channel,
1143 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1144 has also been removed, for the same reason.
1146 * The endcall and enddtmf configuration options are removed. Use the
1147 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1148 channel before calling AgentLogin.
1152 * chan_bridge has been removed. Its functionality has been incorporated
1153 directly into the ConfBridge application itself.
1157 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1158 of the specified span and its B-channels. Note that this command should
1159 only be used if you understand the risks it entails.
1161 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1162 A range of channels can be specified to be destroyed. Note that this command
1163 should only be used if you understand the risks it entails.
1165 * Added the CLI command 'dahdi create channels'. A range of channels can be
1166 specified to be created, or the keyword 'new' can be used to add channels
1169 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1170 the exact configured mailbox name. For app_voicemail mailboxes this is
1173 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1177 * IPv6 support has been added. We are now able to bind to and
1178 communicate using IPv6 addresses.
1182 * The /b option has been removed.
1184 * chan_local moved into the system core and is no longer a loadable module.
1188 * Added general support for busy detection.
1190 * Added ECAM command support for Sony Ericsson phones.
1194 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1195 SIP stack. A collection of resource modules provides the bulk of the SIP
1196 functionality. For more information on the new SIP channel driver, see
1197 https://wiki.asterisk.org/wiki/x/JYGLAQ
1201 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1202 using the 'supportpath' setting, either on a global basis or on a peer basis.
1203 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1204 set of proxies by using a pre-loaded route-set defined by the Path headers in
1205 the REGISTER request. See Realtime updates for more configuration information.
1207 * The SIP_CODEC family of variables may now specify more than one codec. Each
1208 codec must be separated by a comma. The first codec specified is the
1209 preferred codec for the offer. This allows a dialplan writer to specify both
1210 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1212 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1213 in the core, and can be filtered out using the 'eventfilter' parameter
1216 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1217 codecs configured for a peer instead of the requested codec.
1219 * The option "register_retry_403" has been added to chan_sip to work around
1220 servers that are known to erroneously send 403 in response to valid
1221 REGISTER requests and allows Asterisk to continue attepmting to connect.
1225 * Added the 'immeddialkey' parameter. If set, when the user presses the
1226 configured key the already entered number will be immediately dialed. This
1227 is useful when the dialplan allows for variable length pattern matching.
1228 Valid options are '*' and '#'.
1230 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1231 milliseconds) before a call forward is considered to not be answered.
1233 * The 'serviceurl' parameter allows Service URLs to be attached to line
1242 * The password option has been disabled, as the AgentLogin application no
1243 longer provides authentication.
1247 * Due to changes in the Asterisk core, this function is no longer needed to
1248 preserve a MixMonitor on a channel during transfer operations and dialplan
1249 execution. It is effectively obsolete.
1253 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1254 deprecated. Use the CHANNEL function instead to access these attributes.
1256 * The 'l' option has been removed. When reading a CDR attribute, the most
1257 recent record is always used. When writing a CDR attribute, all non-finalized
1260 * The 'r' option has been removed, for the same reason as the 'l' option.
1262 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1267 * A new function CDR_PROP has been added. This function lets you set properties
1268 on a channel's active CDRs. This function is write-only. Properties accept
1269 boolean values to set/clear them on the channel's CDRs. Valid properties
1271 - 'party_a' - make this channel the preferred Party A in any CDR between two
1272 channels. If two channels have this property set, the creation time of the
1273 channel is used to determine who is Party A. Note that dialed channels are
1274 never Party A in a CDR.
1275 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1276 application when set to True, and analogous to the 'e' option in ResetCDR
1281 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1282 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1283 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1286 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1287 string, i.e., [[context],extension],priority. If set on a channel, if a
1288 channel leaves a bridge but is not hung up it will resume dialplan execution
1293 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1294 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1295 The value of this setting is ignored when disabled is used for the argument.
1299 * A new function provided by chan_pjsip, this function can be used in
1300 conjunction with the Dial application to construct a dial string that will
1301 dial all contacts on an Address of Record associated with a chan_pjsip
1306 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1307 outbound channel prior to dialing.
1311 * Redirecting reasons can now be set to arbitrary strings. This means
1312 that the REDIRECTING dialplan function can be used to set the redirecting
1313 reason to any string. It also allows for custom strings to be read as the
1314 redirecting reason from SIP Diversion headers.
1318 * The SPEECH_ENGINE function now supports read operations. When read from, it
1319 will return the current value of the requested attribute.
1323 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1324 system as mailbox@context. The rest of the system cannot add @default
1325 to mailbox identifiers for app_voicemail that do not specify a context
1326 any longer. It is a mailbox identifier format that should only be
1327 interpreted by app_voicemail.
1333 res_agi (Asterisk Gateway Interface)
1335 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1337 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1340 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1341 will start the playback of the audio at the position specified. It will
1342 also return the final position of the file in 'endpos'.
1344 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1345 channel variable if the user stopped the file playback or if a remote
1346 entity stopped the playback. If neither stopped the playback, it will
1347 indicate the overall success/failure of the playback. If stopped early,
1348 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1351 * The SAY ALPHA command now accepts an additional parameter to control
1352 whether it specifies the case of uppercase, lowercase, or all letters to
1353 provide functionality similar to SayAlphaCase.
1355 res_ari (Asterisk RESTful Interface) (and others)
1357 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1358 control telephony primitives in Asterisk by remote client. This includes
1359 channels, bridges, endpoints, media, and other fundamental concepts. Users
1360 of ARI can develop their own communications applications, controlling
1361 multiple channels using an HTTP RESTful interface and receiving JSON events
1362 about the objects via a WebSocket connection. ARI can be configured in
1363 Asterisk via ari.conf. For more information on ARI, see
1364 https://wiki.asterisk.org/wiki/x/0YCLAQ
1368 * Parking has been extracted from the Asterisk core as a loadable module,
1369 res_parking. Configuration for parking is now provided by res_parking.conf.
1370 Configuration through features.conf is no longer supported.
1372 * res_parking uses the configuration framework. If an invalid configuration is
1373 supplied, res_parking will fail to load or fail to reload. Previously,
1374 invalid configurations would generally be accepted, with certain errors
1375 resulting in individually disabled parking lots.
1377 * Parked calls are now placed in bridges. While this is largely an
1378 architectural change, it does have implications on how channels in a parking
1379 lot are viewed. For example, commands that display channels in bridges will
1380 now also display the channels in a parking lot.
1382 * The order of arguments for the new parking applications have been modified.
1383 Timeout and return context/exten/priority are now implemented as options,
1384 while the name of the parking lot is now the first parameter. See the
1385 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1386 in-depth information as well as syntax.
1388 * Extensions are by default no longer automatically created in the dialplan to
1389 park calls or pickup parked calls. Generation of dialplan extensions can be
1390 enabled using the 'parkext' configuration option.
1392 * ADSI functionality for parking is no longer supported. The 'adsipark'
1393 configuration option has been removed as a result.
1395 * The PARKINGSLOT channel variable has been deprecated in favor of
1396 PARKING_SPACE to match the naming scheme of the new system.
1398 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1399 channel even when the configuration option 'comebactoorigin' is enabled.
1401 * A new CLI command 'parking show' has been added. This allows a user to
1402 inspect the parking lots that are currently in use.
1403 'parking show <parkinglot>' will also show the parked calls in a specific
1406 * The CLI command 'parkedcalls' is now deprecated in favor of
1407 'parking show <parkinglot>'.
1409 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1410 can be used to get a list of parked calls for a specific parking lot.
1412 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1413 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1414 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1415 longer a required argument.
1417 * The ParkAndAnnounce application is now provided through res_parking instead
1418 of through the separate app_parkandannounce module.
1420 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1421 by default. Instead, it will follow the timeout rules of the parking lot. The
1422 old behavior can be reproduced by using the 'c' option.
1424 * Dynamic parking lots will now fail to be created under the following
1426 - if the parking lot specified by PARKINGDYNAMIC does not exist
1427 - if they require exclusive park and parkedcall extensions which overlap
1428 with existing parking lots.
1430 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1431 currently contain no calls. Dynamic parking lots containing parked calls
1432 will persist through the reloads without alteration.
1434 * If 'parkext_exclusive' is set for a parking lot and that extension is
1435 already in use when that parking lot tries to register it, this is now
1436 considered a parking system configuration error. Configurations which do
1437 this will be rejected.
1439 * Added channel variable PARKER_FLAT. This contains the name of the extension
1440 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1441 comebacktoorigin is disabled, but the dialplan or an external control
1442 mechanism wants to use the extension in the park-dial context that was
1443 generated to re-dial the parker on timeout.
1445 res_pjsip (and many others)
1447 * A large number of resource modules make up the SIP stack based on pjsip.
1448 The chan_pjsip channel driver users these resource modules to provide
1449 various SIP functionality in Asterisk. The majority of configuration for
1450 these modules is performed in pjsip.conf. Other modules may use their
1451 own configuration files.
1453 * Added 'set_var' option for an endpoint. For each variable specified that
1454 variable gets set upon creation of a channel involving the endpoint.
1458 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1459 them, an Asterisk-specific version of PJSIP needs to be installed.
1460 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1462 res_statsd/res_chan_stats
1464 * A new resource module, res_statsd, has been added, which acts as a statsd
1465 client. This module allows Asterisk to publish statistics to a statsd
1466 server. In conjunction with res_chan_stats, it will publish statistics about
1467 channels to the statsd server. It can be configured via res_statsd.conf.
1471 * Device state for XMPP buddies is now available using the following format:
1472 XMPP/<client name>/<buddy address>
1473 If any resource is available the device state is considered to be not in use.
1474 If no resources exist or all are unavailable the device state is considered
1481 Realtime/Database Scripts
1483 * Asterisk previously included example db schemas in the contrib/realtime/
1484 directory of the source tree. This has been replaced by a set of database
1485 migrations using the Alembic framework. This allows you to use alembic to
1486 initialize the database for you. It will also serve as a database migration
1487 tool when upgrading Asterisk in the future.
1489 See contrib/ast-db-manage/README.md for more details.
1493 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1494 This python script will convert an existing sip.conf file to a
1495 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1496 is meant to be an aid in converting an existing chan_sip configuration to
1497 a chan_pjsip configuration, but it is expected that configuration beyond
1498 what the script provides will be needed.
1501 ------------------------------------------------------------------------------
1502 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1503 ------------------------------------------------------------------------------
1507 * The Asterisk build system will now build and install a shared library
1508 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1509 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1510 that Asterisk can ensure that these functions do *not* get called by any
1511 modules that are loaded into Asterisk, since they should only be called once
1512 in any single process. If desired, this feature can be disabled by supplying
1513 the "--disable-asteriskssl" option to the configure script.
1515 * A new make target, 'full', has been added to the Makefile. This performs
1516 the same compilation actions as make all, but will also scan the entirety of
1517 each source file for documentation. This option is needed to generate AMI
1518 event documentation. Note that your system must have Python in order for
1519 this make target to succeed.
1521 * The optimization portion of the build system has been reworked to avoid
1522 broken builds on certain architectures. All architecture-specific
1523 optimization has been removed in favor of using -march=native to allow gcc
1524 to detect the environment in which it is running when possible. This can
1525 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1527 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1528 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1530 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1531 previously parsed the header file to obtain the version of Asterisk, you
1532 will now have to go through Asterisk to get the version information.
1540 * Added 'F()' option. Similar to the dial option, this can be supplied with
1541 arguments indicating where the callee should go after the caller is hung up,
1542 or without options specified, the priority after the Queue will be used.
1547 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1548 all non-admin participants on a conference. The confbridge configuration
1549 file also allows for the default sounds played to all conference users when
1550 this occurs to be overriden using sound_participants_unmuted and
1551 sound_participants_muted.
1553 * Added menu action participant_count. This will playback the number of
1554 current participants in a conference.
1556 * Added announcement configuration option to user profile. If set the sound
1557 file will be played to the user, and only the user, upon joining the
1560 * Added record_file_append option that defaults to "yes", but if set to no
1561 will create a new file between each start/stop recording.
1566 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1567 channels respectively before the callee channels are called.
1572 * Added support for IPv6.
1574 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1575 external process will cause the current playlist to be cleared, including
1576 stopping any audio file that is currently playing. This is useful when you
1577 want to interrupt audio playback only when specific DTMF is entered by the
1583 * A new option, 'I' has been added to app_followme. By setting this option,
1584 Asterisk will not update the caller with connected line changes when they
1585 occur. This is similar to app_dial and app_queue.
1587 * The 'N' option is now ignored if the call is already answered.
1589 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1590 and caller channels respectively before the callee channels are called.
1592 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1598 * MixMonitor hooks now have IDs associated with them which can be used to
1599 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1600 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1601 now accepts that ID as an argument.
1603 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1604 indicated mailboxes.
1609 * The connect action in app_mysql now allows you to specify a port number to
1610 connect to. This is useful if you run a MySQL server on a non-standard
1616 * Increased the default number of allowed destinations from 5 to 12.
1621 * The app_page application now no longer depends on DAHDI or app_meetme. It
1622 has been re-architected to use app_confbridge internally.
1627 * Added queue options autopausebusy and autopauseunavail for automatically
1628 pausing a queue member when their device reports busy or congestion.
1630 * The 'ignorebusy' option for queue members has been deprecated in favor of
1631 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1632 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1633 per interface basis. Individual ringinuse values can now be set in
1634 queues.conf via an argument to member definitions. Lastly, the queue
1635 'ringinuse' setting now only determines defaults for the per member
1636 'ringinuse' setting and does not override per member settings like it does
1637 in earlier versions.
1639 * Added 'F()' option. Similar to the dial option, this can be supplied with
1640 arguments indicating where the callee should go after the caller is hung up,
1641 or without options specified, the priority after the Queue will be used.
1643 * Added new option log_member_name_as_agent, which will cause the membername to
1644 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1645 state_interface has been set.
1647 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1649 * App_queue will now play periodic announcements for the caller that
1650 holds the first position in the queue while waiting for answer.
1654 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1655 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1656 changed arguments to SayUnixTime so that every option is truly optional even
1657 when using multiple options (so that j option could be used without having to
1658 manually specify timezone and format) There are other benefits, e.g., format
1659 can now be used without specifying time zone as well.
1664 * Addition of the VM_INFO function - see Function changes.
1666 * The imapserver, imapport, and imapflags configuration options can now be
1667 overriden on a user by user basis.
1669 * When voicemail plays a message's envelope with saycid set to yes, when
1670 reaching the caller id field it will play a recording of a file with the same
1671 base name as the sender's callerid if there is a similarly named file in
1672 <astspooldir>/recordings/callerids/
1674 * Voicemails now contains a unique message identifier "msg_id", which is stored
1675 in the message envelope with the sound files. IMAP backends will now store
1676 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1677 backends will store the message identifier in a "msg_id" column. See
1678 UPGRADE.txt for more information.
1680 * Added VoiceMailPlayMsg application. This application will play a single
1681 voicemail message from a mailbox. The result of the application, SUCCESS or
1682 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1687 * Hangup handlers can be attached to channels using the CHANNEL() function.
1688 Hangup handlers will run when the channel is hung up similar to the h
1689 extension. The hangup_handler_push option will push a GoSub compatible
1690 location in the dialplan onto the channel's hangup handler stack. The
1691 hangup_handler_pop option will remove the last added location, and optionally
1692 replace it with a new GoSub compatible location. The hangup_handler_wipe
1693 option will remove all locations on the stack, and optionally add a new
1696 * The expression parser now recognizes the ABS() absolute value function,
1697 which will convert negative floating point values to positive values.
1699 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1700 control of faxdetect.
1702 * Addition of the VM_INFO function that can be used to retrieve voicemail
1703 user information, such as the email address and full name.
1704 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1707 * The REDIRECTING function now supports the redirecting original party id
1710 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1711 lets you set some of the configuration options from the [general] section
1712 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1713 the key sequence used to activate built-in features, such as blindxfer,
1714 and automon. See the built-in documentation for details.
1716 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1717 instead of simply the uri. This is the format that MessageSend() can use
1718 in the from parameter for outgoing SIP messages.
1720 * Added the PRESENCE_STATE function. This allows retrieving presence state
1721 information from any presence state provider. It also allows setting
1722 presence state information from a CustomPresence presence state provider.
1723 See AMI/CLI changes for related commands.
1725 * Added the AMI_CLIENT function to make manager account attributes available
1726 to the dialplan. It currently supports returning the current number of
1727 active sessions for a given account.
1729 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1730 and the REDIRECTING functions.
1738 * Added a manager event "LocalBridge" for local channel call bridges between
1739 the two pseudo-channels created.
1744 * Added dialtone_detect option for analog ports to disconnect incoming
1745 calls when dialtone is detected.
1747 * Added option colp_send to send ISDN connected line information. Allowed
1748 settings are block, to not send any connected line information; connect, to
1749 send connected line information on initial connect; and update, to send
1750 information on any update during a call. Default is update.
1752 * Add options namedcallgroup and namedpickupgroup to support installations
1753 where a higher number of groups (>64) is required.
1755 * Added support to use private party ID information with PRI calls.
1760 * A new channel driver named chan_motif has been added which provides support for
1761 Google Talk and Jingle in a single channel driver. This new channel driver includes
1762 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1763 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1764 specification, current Google Jingle specification, and the original Google Talk
1770 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1771 globally and overriden on a peer by peer basis.
1773 * Direct media functionality has been added. Options in config are:
1774 directmedia (directrtp) and directrtpsetup (earlydirect)
1776 * ChannelUpdate events now contain a CallRef header.
1781 * Asterisk will no longer substitute CID number for CID name in the display
1782 name field if CID number exists without a CID name. This change improves
1783 compatibility with certain device features such as Avaya IP500's directory
1786 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1787 created using that setting to not be removed during SIP reload.
1789 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1790 request with a "Record:" header, this will turn the requested feature on/off.
1791 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1792 dynamic features must be enabled and configured properly on the requesting
1793 channel for this to function properly.
1795 * Add support to realtime for the 'callbackextension' option.
1797 * When multiple peers exist with the same address, but differing
1798 callbackextension options, incoming requests that are matched by address
1799 will be matched to the peer with the matching callbackextension if it is
1802 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1803 which set the force_rport and comedia options automatically if Asterisk
1804 detects that an incoming SIP request crossed a NAT after being sent by
1805 the remote endpoint.
1807 * The default global nat setting in sip.conf has been changed from force_rport
1808 to auto_force_rport.
1810 * NAT settings are now a combinable list of options. The equivalent of the
1811 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1813 * Adds an option send_diversion which can be disabled to prevent
1814 diversion headers from automatically being added to INVITE requests.
1816 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1817 be sent to the remote host at a given interval to keep the NAT mapping open.
1818 This can be enabled using the keepalive configuration option.
1820 * Add option 'tonezone' to specify country code for indications. This option
1821 can be set both globally and overridden for specific peers.
1823 * The SIP Security Events Framework now supports IPv6.
1825 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1826 between multiple user agents. When set, for directmedia reinvites,
1827 Asterisk will not send an immediate reinvite on an incoming call leg. This
1828 option is useful when peered with another SIP user agent that is known to
1829 send immediate direct media reinvites upon call establishment.
1831 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1834 * Add options subminexpiry and submaxexpiry to set limits of subscription
1835 timer independently from registration timer settings. The setting of the
1836 registration timer limits still is done by options minexpiry, maxexpiry
1837 and defaultexpiry. For backwards compatibility the setting of minexpiry
1838 and maxexpiry also is used to configure the subscription timer limits if
1839 subminexpiry and submaxexpiry are not set in sip.conf.
1841 * Set registration timer limits to default values when reloading sip
1842 configuration and values are not set by configuration.
1844 * Add options namedcallgroup and namedpickupgroup to support installations
1845 where a higher number of groups (>64) is required.
1847 * When a MESSAGE request is received, the address the request was received from
1848 is now saved in the SIP_RECVADDR variable.
1850 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1851 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1852 the ANI2/OLI information is set on the channel, which can be retrieved using
1853 the CALLERID function.
1855 * Peers can now be configured to support negotiation of ICE candidates using
1856 the setting icesupport. See res_rtp_asterisk changes for more information.
1858 * Added support for format attribute negotiation. See the Codecs changes for
1861 * Extra headers specified with SIPAddHeader are sent with the REFER message
1862 when using Transfer application. See refer_addheaders in sip.conf.sample.
1864 * Added support to use private party ID information with calls.
1866 * Adds an option discard_remote_hold_retrieval that when set stops telling
1867 the peer to start music on hold.
1872 * Added skinny version 17 protocol support.
1876 --------------------
1877 * Added ability to use multiple lines for a single phone. This allows multiple
1878 calls to occur on a single phone, using callwaiting and switching between calls.
1880 * Added option 'sharpdial' allowing end dialing by pressing # key
1882 * Added option 'interdigit_timer' to control phone dial timeout
1884 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1886 * Added global 'debug' option, that enables debug in channel driver
1888 * Added ability to translate on-screen menu in multiple languages. Tested on
1889 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1890 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1893 * In addition to English added French and Russian languages for on-screen menus
1895 * Reworked dialing number input: added dialing by timeout, immediate dial on
1896 on dialplan compare, phone number length now not limited by screen size
1898 * Added ability to pickup a call using features.conf defined value and
1904 * Add options namedcallgroup and namedpickupgroup to support installations
1905 where a higher number of groups (>64) is required.
1907 * Added support to use private party ID information with calls.
1912 * The minimum DTMF duration can now be configured in asterisk.conf
1913 as "mindtmfduration". The default value is (as before) set to 80 ms.
1914 (previously it was only available in source code)
1916 * Named ACLs can now be specified in acl.conf and used in configurations that
1917 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1918 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1919 working ACL. In addition, some CLI commands have been added to provide
1920 show information and allow for module reloading - see CLI Changes.
1922 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1923 items (separated by commas), and items in the rule can be negated by prefixing
1924 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1925 longer necessray to control the order that the 'permit' and 'deny' columns are
1926 returned from queries.
1928 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1929 be used within the dynamic weight attribute when specifying a mapping.
1931 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1932 header, instead of putting the user defined event name there. When enabled
1933 the UserDefType header is added for user defined events. This feature is
1934 enabled with the setting show_user_defined.
1936 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1937 line purposes use the following variables instead of their macro equivalents:
1938 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1939 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1940 cc_callback_macro in channel configurations.
1942 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1945 * Call files now support the "early_media" option to connect with an outgoing
1946 extension when early media is received.
1948 * Added support to use private party ID information with calls.
1953 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1954 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1955 AGI application would exit immediately after a channel hangup is detected.
1957 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1958 are resolved and each address is attempted in turn until one succeeds or
1962 AMI (Asterisk Manager Interface)
1964 * The originate action now has an option "EarlyMedia" that enables the
1965 call to bridge when we get early media in the call. Previously,
1966 early media was disregarded always when originating calls using AMI.
1968 * Added setvar= option to manager accounts (much like sip.conf)
1970 * Originate now generates an error response if the extension given is not found
1973 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1974 them if the i(variable) option is used. StopMixMonitor will accept
1975 MixMonitorID as an option to close specific MixMonitors.
1977 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1978 updated to include information about peers configured with
1979 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1980 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1981 returned if auto_force_rport is not enabled.
1983 * Added SIPpeerstatus manager command which will generate PeerStatus events
1984 similar to the existing PeerStatus events found in chan_sip on demand.
1986 * Hangup now can take a regular expression as the Channel option. If you want
1987 to hangup multiple channels, use /regex/ as the Channel option. Existing
1988 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1989 the manager will send you a list of channels back that were hung up.
1991 * Support for IPv6 addresses has been added.
1993 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1994 documentation is only generated when Asterisk is compiled using 'make full'.
1995 See the CLI section for commands to display AMI event information.
1997 * The AMI Hangup event now includes the AccountCode header so you can easily
1998 correlate with AMI Newchannel events.
2000 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2001 the StateInterface of the queue member.
2003 * Added AMI event SessionTimeout in the Call category that is issued when a
2004 call is terminated due to either RTP stream inactivity or SIP session timer
2007 * CEL events can now contain a user defined header UserDefType. See core
2008 changes for more information.
2010 * OOH323 ChannelUpdate events now contain a CallRef header.
2012 * Added PresenceState command. This command will report the presence state for
2013 the given presence provider.
2015 * Added Parkinglots command. This will list all parking lots as a series of
2016 AMI Parkinglot events.
2018 * Added MessageSend command. This behaves in the same manner as the
2019 MessageSend application, and is a technolgoy agnostic mechanism to send out
2020 of call text messages.
2022 * Added "message" class authorization. This grants an account permission to
2023 send out of call messages. Write-only.
2028 * The "dialplan add include" command has been modified to create context a context
2029 if one does not already exist. For instance, "dialplan add include foo into bar"
2030 will create context "bar" if it does not already exist.
2032 * A "dialplan remove context" command has been added to remove a context from
2035 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2036 filenames of all running mixmonitors on a channel.
2038 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2039 numeric instead of 0, 1, or 2.
2041 * "stun show status" will show a table describing how the STUN client is
2044 * "acl show [named acl]" will show information regarding a Named ACL. The
2045 acl module can be reloaded with "reload acl".
2047 * Added CLI command to display AMI event information - "manager show events",
2048 which shows a list of all known and documented AMI events, and "manager show
2049 event [event name]", which shows detail information about a specific AMI
2052 * The result of the CLI command "queue show" now includes the state interface
2053 information of the queue member.
2055 * The command "core set verbose" will now set a separate level of logging for
2056 each remote console without affecting any other console.
2058 * Added command "cdr show pgsql status" to check connection status
2060 * "sip show channel" will now display the complete route set.
2062 * Added "presencestate list" command. This command will list all custom
2063 presence states that have been set by using the PRESENCE_STATE dialplan
2066 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2067 command. This changes a custom presence to a new state.
2072 * Codec lists may now be modified by the '!' character, to allow succinct
2073 specification of a list of codecs allowed and disallowed, without the
2074 requirement to use two different keywords. For example, to specify all
2075 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2077 * Add support for parsing SDP attributes, generating SDP attributes, and
2078 passing it through. This support includes codecs such as H.263, H.264, SILK,
2079 and CELT. You are able to set up a call and have attribute information pass.
2080 This should help considerably with video calls.
2082 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2083 just like the GSM codec.
2087 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2088 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2092 * Asterisk version and build information is now logged at the beginning of a
2095 * Threads belonging to a particular call are now linked with callids which get
2096 added to any log messages produced by those threads. Log messages can now be
2097 easily identified as involved with a certain call by looking at their call id.
2098 Call ids may also be attached to log messages for just about any case where
2099 it can be determined to be related to a particular call.
2101 * Each logging destination and console now have an independent notion of the
2102 current verbosity level. Logger.conf now allows an optional argument to
2103 the 'verbose' specifier, indicating the level of verbosity sent to that
2104 particular logging destination. Additionally, remote consoles now each
2105 have their own verbosity level. The command 'core set verbose' will now set
2106 a separate level for each remote console without affecting any other
2112 * Added 'announcement' option which will play at the start of MOH and between
2113 songs in modes of MOH that can detect transitions between songs (eg.
2119 * New per parking lot options: comebackcontext and comebackdialtime. See
2120 configs/features.conf.sample for more details.
2122 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2125 * Channel variable PARKEDCALL is now set with the name of the parking lot
2126 when a timeout occurs.
2132 CDR Postgresql Driver
2134 * Added command "cdr show pgsql status" to check connection status
2137 CDR Adaptive ODBC Driver
2139 * Added schema option for databases that support specifying a schema.
2147 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2148 CALENDAR_WRITE has completed successfully.
2153 * A new option, 'probation' has been added to rtp.conf
2154 RTP in strictrtp mode can now require more than 1 packet to exit learning
2155 mode with a new source (and by default requires 4). The probation option
2156 allows the user to change the required number of packets in sequence to any
2157 desired value. Use a value of 1 to essentially restore the old behavior.
2158 Also, with strictrtp on, Asterisk will now drop all packets until learning
2159 mode has successfully exited. These changes are based on how pjmedia handles
2160 media sources and source changes.
2162 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2163 enabled or disabled using the icesupport setting. A variety of other
2164 settings have been introduced to configure STUN/TURN connections.
2169 * A new module, res_corosync, has been introduced. This module uses the
2170 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2171 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2172 Device State (presence) information. This module is very similar to, and
2173 is a replacement for the res_ais module that was in previous releases of
2179 * This module adds a cleaned up, drop-in replacement for res_jabber called
2180 res_xmpp. This provides the same externally facing functionality but is
2181 implemented differently internally. res_jabber has been deprecated in favor
2182 of res_xmpp; please see the UPGRADE.txt file for more information.
2187 * The safe_asterisk script has been updated to allow several of its parameters
2188 to be set from environment variables. This also enables a custom run
2189 directory of Asterisk to be specified, instead of defaulting to /tmp.
2191 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2192 its value to determine the directory to assume is the top-level directory of
2193 the source tree. If the variable is not set, it defaults to the current
2194 behavior and uses the current working directory.
2196 ------------------------------------------------------------------------------
2197 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2198 ------------------------------------------------------------------------------
2202 * Asterisk now has protocol independent support for processing text messages
2203 outside of a call. Messages are routed through the Asterisk dialplan.
2204 SIP MESSAGE and XMPP are currently supported. There are options in
2205 jabber.conf and sip.conf to allow enabling these features.
2206 -> jabber.conf: see the "sendtodialplan" and "context" options.
2207 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2208 and "outofcall_message_context" options.
2209 The MESSAGE() dialplan function and MessageSend() application have been
2210 added to go along with this functionality. More detailed usage information
2211 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2212 * If real-time text support (T.140) is negotiated, it will be preferred for
2213 sending text via the SendText application. For example, via SIP, messages
2214 that were once sent via the SIP MESSAGE request would be sent via RTP if
2215 T.140 text is negotiated for a call.
2219 * parkedmusicclass can now be set for non-default parking lots.
2221 Asterisk Manager Interface
2222 --------------------------
2223 * PeerStatus now includes Address and Port.
2224 * Added Hold events for when the remote party puts the call on and off hold
2225 for chan_dahdi ISDN channels.
2226 * Added new action MeetmeListRooms to list active conferences (shows same
2227 data as "meetme list" at the CLI).
2228 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2229 Description field that is set by 'description' in the channel configuration
2231 * Added Uniqueid header to UserEvent.
2232 * Added new action FilterAdd to control event filters for the current session.
2233 This requires the system permission and uses the same filter syntax as
2234 filters that can be defined in manager.conf
2235 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2236 versions had some instances of the event converted, but others were left
2237 as-is. All Unlink events should now be converted to Bridge events. The AMI
2238 protocol version number was incremented to 1.2 as a result of this change.
2240 Asterisk HTTP Server
2241 --------------------------
2242 * The HTTP Server can bind to IPv6 addresses.
2245 --------------------------
2246 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2247 with busydetect. usage example: busypattern=200,200,200,600
2250 --------------------------
2251 * New 'gtalk show settings' command showing the current settings loaded from
2253 * The 'logger reload' command now supports an optional argument, specifying an
2254 alternate configuration file to use.
2255 * 'dialplan add extension' command will now automatically create a context if
2256 the specified context does not exist with a message indicated it did so.
2257 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2258 Description field which can be populated with 'description' in the channel
2259 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2262 --------------------------
2263 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2264 thus allowing records which do NOT match the specified filter.
2265 * Added ability to log CONGESTION calls to CDR
2268 --------------------------
2269 * Ability to define custom SILK formats in codecs.conf.
2270 * Addition of speex32 audio format with translation.
2271 * CELT codec pass-through support and ability to define
2272 custom CELT formats in codecs.conf.
2273 * Ability to read raw signed linear files with sample rates
2274 ranging from 8khz - 192khz. The new file extensions introduced
2275 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2276 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2277 Skinny, H.323, etc) can still only support the following codecs:
2278 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2279 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2280 Video: h261, h263, h263p, h264, mpeg4
2285 --------------------------
2286 * New highly optimized and customizable ConfBridge application capable of
2287 mixing audio at sample rates ranging from 8khz-96khz.
2288 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2289 and bridge profiles on a channel.
2290 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2291 about a conference such as locked status and number of parties, admins,
2293 * Addition of video_mode option in confbridge.conf for adding video support
2294 into a bridge profile.
2295 * Addition of the follow_talker video_mode in confbridge.conf. This video
2296 mode dynamically switches the video feed to always display the loudest talker
2297 supplying video in the conference.
2301 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2302 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2303 variables from asterisk.conf.
2307 * Addition of the JITTERBUFFER dialplan function. This function allows
2308 for jitterbuffering to occur on the read side of a channel. By using
2309 this function conference applications such as ConfBridge and MeetMe can
2310 have the rx streams jitterbuffered before conference mixing occurs.
2311 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2313 * Added STRREPLACE function. This function let's the user search a variable
2314 for a given string to replace with another string as many times as the
2315 user specifies or just throughout the whole string.
2316 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2317 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2318 * Added extensions to chan_ooh323 in function CHANNEL()
2320 libpri channel driver (chan_dahdi) DAHDI changes
2321 --------------------------
2322 * Added moh_signaling option to specify what to do when the channel's bridged
2323 peer puts the ISDN channel on hold.
2324 * Added display_send and display_receive options to control how the display ie
2325 is handled. To send display text from the dialplan use the SendText()
2326 application when the option is enabled.
2327 * Added mcid_send option to allow sending a MCID request on a span.
2330 --------------------------
2331 * Added setvar option to calendar.conf to allow setting channel variables on
2332 notification channels.
2333 * Added "calendar show types" CLI command to list registered calendar
2337 --------------------------
2338 * Added two new options, r and t with file name arguments to record
2339 single direction (unmixed) audio recording separate from the bidirectional
2340 (mixed) recording. The mixed file name argument is optional now as long
2341 as at least one recording option is used.
2344 --------------------------
2345 * Added a new option, l, which will disable local call optimization for
2346 channels involved with the FollowMe thread. Use this option to improve
2347 compatability for a FollowMe call with certain dialplan apps, options, and
2351 --------------------------
2352 * Added option "k" that will automatically close the conference when there's
2353 only one person left when a user exits the conference.
2356 --------------------------
2357 * cel_pgsql now supports the 'extra' column for data added using the
2358 CELGenUserEvent() application.
2361 --------------------------
2362 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2363 in the sample extensions.lua file for syntax details.
2364 * Applications that perform jumps in the dialplan such as Goto will now
2365 execute properly. When pbx_lua detects that the context, extension, or
2366 priority we are executing on has changed it will immediately return control
2367 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2368 the priority after the currently executing priority.
2369 * An autoservice is now started by default for pbx_lua channels. It can be
2370 stopped and restarted using the autoservice_stop() and autoservice_start()
2374 --------------------------
2375 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2376 into a FAXStatus event with an 'Operation' header that will be either
2377 'send', 'receive', and 'gateway'.
2378 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2379 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2380 feature will handle converting a fax call between an audio T.30 fax terminal
2381 and an IFP T.38 fax terminal.
2385 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2386 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2387 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2391 * Added general option negative_penalty_invalid default off. when set
2392 members are seen as invalid/logged out when there penalty is negative.
2393 for realtime members when set remove from queue will set penalty to -1.
2394 * Added queue option autopausedelay when autopause is enabled it will be
2395 delayed for this number of seconds since last successful call if there
2396 was no prior call the agent will be autopaused immediately.
2397 * Added member option ignorebusy this when set and ringinuse is not
2398 will allow per member control of multiple calls as ringinuse does for
2403 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2405 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2406 one participant left (much like a normal call bridge)
2407 * Added extra argument to Originate to set timeout.
2411 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2412 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2413 utility in the UTILS section of menuselect. If an existing astdb is found and no
2414 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2415 convert an existing astdb to the SQLite3 version automatically at runtime.
2419 * Modules marked as deprecated are no longer marked as building by default. Enabling
2420 these modules is still available via menuselect.
2424 * authdebug is now disabled by default. To enable this functionaility again
2425 set authdebug = yes in iax.conf.
2429 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2430 releases it was disabled.
2434 * The PBX core previously made a call with a non-existing extension test for
2435 extension s@default and jump there if the extension existed.
2436 This was a bad default behaviour and violated the principle of least surprise.
2437 It has therefore been changed in this release. It may affect some
2438 applications and configurations that rely on this behaviour. Most channel
2439 drivers have avoided this for many releases by testing whether the extension
2440 called exists before starting the PBX and generating a local error.
2441 This behaviour still exists and works as before.
2443 Extension "s" is used when no extension is given in a channel driver,
2444 like immediate answer in DAHDI or calling to a domain with no user part
2447 ------------------------------------------------------------------------------
2448 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2449 ------------------------------------------------------------------------------
2453 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2454 now defaults to force_rport. It is very important that phones requiring nat=no be
2455 specifically set as such instead of relying on the default setting. If at all
2456 possible, all devices should have nat settings configured in the general section as
2457 opposed to configuring nat per-device.
2458 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2459 codecs sent in response to an INVITE to the single most preferred codec.
2460 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2461 to be used for the outgoing call. It must be one of the codecs configured
2463 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2464 to be used for holding a private key. If tlsprivatekey is not specified,
2465 tlscertfile is searched for both public and private key.
2466 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2467 outbound client connections to be specified.
2468 * The sendrpid parameter has been expanded to include the options
2469 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2470 header to be sent (equivalent to setting sendrpid=yes) and setting
2471 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2472 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2473 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2474 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2475 will accept the SDP even if the SDP version number is not properly incremented,
2476 but will generate a warning in the log indicating that the SIP peer that sent
2477 the SDP should have the 'ignoresdpversion' option set.
2478 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2479 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2480 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2481 remote side requests it and disables symmetric RTP support. Setting it to
2482 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2483 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2484 and enables symmetric RTP support.
2485 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2486 response. This permits the master channel to know how each channel dialled
2487 in a multi-channel setup resolved in an individual way. This carries a
2488 performance penalty and can be disabled in sip.conf using the
2489 'storesipcause' option.
2490 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2491 configuration for the externip and externhost options when tcp or tls is used.
2492 * Added support for message body (stored in content variable) to SIP NOTIFY message
2493 accessible via AMI and CLI.
2494 * Added 'media_address' configuration option which can be used to explicitly specify
2495 the IP address to use in the SDP for media (audio, video, and text) streams.
2496 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2497 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2499 * Added 'use_q850_reason' configuration option for generating and parsing
2500 if available Reason: Q.850;cause=<cause code> header. It is implemented
2501 in some gateways for better passing PRI/SS7 cause codes via SIP.
2502 * When dialing SIP peers, a new component may be added to the end of the dialstring
2503 to indicate that a specific remote IP address or host should be used when dialing
2504 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2505 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2506 ability to selectively force bridged channels to also be encrypted is also
2507 implemented. Branching in the dialplan can be done based on whether or not
2508 a channel has secure media and/or signaling.
2509 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2511 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2512 Charge messages to snom phones.
2513 * Added support for G.719 media streams.
2514 * Added support for 16khz signed linear media streams.
2515 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2516 RTP has been outfitted with the same abilities.
2517 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2518 available in device configurations as well as in the dial plan.
2519 * Addition of the 'subscribe_network_change' option for turning on and off
2520 res_stun_monitor module support in chan_sip.
2521 * Addition of the 'auth_options_requests' option for turning on and off
2522 authentication for OPTIONS requests in chan_sip.
2526 * Add #tryinclude statement for config files. This provides the same
2527 functionality as the #include statement however an asterisk module will
2528 still load if the filename does not exist. Using the #include statement
2529 Asterisk will not allow the module to load.
2533 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2534 on realtime updates.
2535 * Added the ability for chan_iax2 to inform the dialplan whether or not
2536 encryption is being used. This interoperates with the SIP SRTP implementation
2537 so that a secure SIP call can be bridged to a secure IAX call when the
2538 dialplan requires bridged channels to be "secure".
2539 * Addition of the 'subscribe_network_change' option for turning on and off
2540 res_stun_monitor module support in chan_iax.
2545 * Added ability to preset channel variables on indicated lines with the setvar
2546 configuration option. Also, clearvars=all resets the list of variables back
2548 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2549 See configs/res_pktccops.conf for more information.
2551 XMPP Google Talk/Jingle changes
2552 -------------------------------
2553 * Added the externip option to gtalk.conf.
2554 * Added the stunaddr option to gtalk.conf which allows for the automatic
2555 retrieval of the external ip from a stun server.
2559 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2560 match to a partial channel name.
2561 * Added .m3u support for Mp3Player application.
2562 * Added progress option to the app_dial D() option. When progress DTMF is
2563 present, those values are sent immediately upon receiving a PROGRESS message
2564 regardless if the call has been answered or not.
2565 * Added functionality to the app_dial F() option to continue with execution
2566 at the current location when no parameters are provided.
2567 * Added the 'a' option to app_dial to answer the calling channel before any
2568 announcements or macros are executed.
2569 * Modified app_dial to set answertime when the called channel answers even if
2570 the called channel hangs up during playback of an announcement.
2571 * Modified app_dial 'r' option to support an additional parameter to play an
2572 indication tone from indications.conf
2573 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2574 to cycle through the next available channel. By default this is still '*'.
2575 * Added x() option to app_chanspy. This option allows DTMF to be set to
2576 exit the application.
2577 * The Voicemail application has been improved to automatically ignore messages
2578 that only contain silence.
2579 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2580 associated mailbox(es) to be greetings-only.
2581 * The ChanSpy application now has the 'S' option, which makes the application
2582 automatically exit once it hits a point where no more channels are available
2584 * The ChanSpy application also now has the 'E' option, which spies on a single
2585 channel and exits when that channel hangs up.
2586 * The MeetMe application now turns on the DENOISE() function by default, for
2587 each participant. In our tests, this has significantly decreased background
2588 noise (especially noisy data centers).
2589 * Voicemail now permits storage of secrets in a separate file, located in the
2590 spool directory of each individual user. The control for this is located in
2591 the "passwordlocation" option in voicemail.conf. Please see the sample
2592 configuration for more information.
2593 * The ChanIsAvail application now exposes the returned cause code using a separate
2594 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2595 * Added 'd' option to app_followme. This option disables the "Please hold"
2597 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2598 received will terminate recording.
2599 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2600 Previously the folder could only be set per context, but has now been extended
2601 using the imapfolder option.
2602 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2603 * Voicemail now allows the pager date format to be specified separately from the
2605 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2606 to allow joining, leaving, and sending text to group chats.
2607 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2608 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2609 to all paged phones (and optionally excluding the caller's one using the new
2610 option 'n') before the call is bridged.
2611 * The 'f' option to Dial has been augmented to take an optional argument. If no
2612 argument is provided, the 'f' option works as it always has. If an argument is
2613 provided, then the connected party information of all outgoing channels created
2614 during the Dial will be set to the argument passed to the 'f' option.
2615 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2617 * The OSP lookup application adds in/outbound network ID, optional security,
2618 number portability, QoS reporting, destination IP port, custom info and service
2620 * Added new application VMSayName that will play the recorded name of the voicemail
2621 user if it exists, otherwise will play the mailbox number.
2622 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2623 retrieve state for a particular bridge, where <name> is the conference name
2624 * app_directory now allows exiting at any time using the operator or pound key.
2625 * Voicemail now supports setting a locale per-mailbox.
2626 * Two new applications are provided for declining counting phrases in multiple
2627 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2629 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2631 * Voicemail now includes rdnis within msgXXXX.txt file.
2632 * ExternalIVR now supports IPv6 addresses.
2633 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2634 at https://wiki.asterisk.org/wiki/x/oQBB
2635 * ParkedCall and Park can now specify the parking lot to use.
2639 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2640 over SRV records associated with a specific service. From the CLI, type
2641 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2642 details on how these may be used.
2643 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2644 pitch of a channel's tx and rx audio streams.
2645 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2646 setting various connected line and redirecting party information.
2647 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2648 support ISDN subaddressing.
2649 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2650 * For DAHDI channels, the CHANNEL() dialplan function now allows
2651 the dialplan to request changes in the configuration of the active
2652 echo canceller on the channel (if any), for the current call only.
2655 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2657 The possible values are:
2659 on - normal mode (the echo canceller is actually reinitialized)
2661 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2663 voice - voice mode (returns from FAX mode, reverting the changes that
2664 were made when FAX mode was requested)
2665 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2666 and setting variables on the channel which created the current channel.
2667 Administrators should take care to avoid naming conflicts, when multiple
2668 channels are dialled at once, especially when used with the Local channel
2669 construct (which all could set variables on the master channel). Usage
2670 of the HASH() dialplan function, with the key set to the name of the slave
2671 channel, is one approach that will avoid conflicts.
2672 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2674 * func_odbc now allows multiple row results to be retrieved without using
2675 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2676 from the same query by using the name of the function which retrieved the
2677 first row as an argument to ODBC_FETCH().
2678 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2679 dialplan. This function returns the content of the received message.
2680 * Added REPLACE, which searches a given variable name for a set of characters,
2681 then either replaces them with a single character or deletes them.
2682 * Added PASSTHRU, which literally passes the same argument back as its return
2683 value. The intent is to be able to use a literal string argument to
2684 functions that currently require a variable name as an argument.
2685 * HASH-associated variables now can be inherited across channel creation, by
2686 prefixing the name of the hash at assignment with the appropriate number of
2687 underscores, just like variables.
2688 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2689 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2690 whether or not channels that are bridged to the current channel will be
2691 required to have secure signaling and/or media.
2692 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2693 the current channel has secure signaling and/or media.
2694 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2695 "no_media_path" option.
2696 Returns "0" if there is a B channel associated with the call.
2697 Returns "1" if no B channel is associated with the call. The call is either
2698 on hold or is a call waiting call.
2699 * Added option to dialplan function CDR(), the 'f' option
2700 allows for high resolution times for billsec and duration fields.
2701 * FILE() now supports line-mode and writing.
2702 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2703 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2707 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2708 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2709 and is set when a dynamic feature is triggered.
2710 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2711 to dynamically create a new parking lot matching the value this varible is
2713 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2714 features.conf that should be the base for dynamic parkinglots.
2715 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2716 parkinglot should have.
2717 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2718 parkinglot should have.
2719 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2724 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2725 timeout has expired.
2726 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2727 to the caller when an Agent's phone is ringing. This can be used to indicate
2728 to the caller that their call is about to be picked up, which is nice when
2729 one has been on hold for an extened period of time.
2730 * A new config option, penaltymemberslimit, has been added to queues.conf.
2731 When set this option will disregard penalty settings when a queue has too
2733 * A new option, 'I' has been added to both app_queue and app_dial.
2734 By setting this option, Asterisk will not update the caller with
2735 connected line changes or redirecting party changes when they occur.
2736 * A 'relative-periodic-announce' option has been added to queues.conf. When
2737 enabled, this option will cause periodic announce times to be calculated
2738 from the end of announcements rather than from the beginning.
2739 * The autopause option in queues.conf can be passed a new value, "all." The
2740 result is that if a member becomes auto-paused, he will be paused in all
2741 queues for which he is a member, not just the queue that failed to reach
2743 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2744 * The queue logger now allows events to optionally propagate to a file,
2745 even when realtime logging is turned on. Additionally, realtime logging
2746 supports sending the event arguments to 5 individual fields, although it
2747 will fallback to the previous data definition, if the new table layout is
2750 mISDN channel driver (chan_misdn) changes
2751 ----------------------------------------
2752 * Added display_connected parameter to misdn.conf to put a display string
2753 in the CONNECT message containing the connected name and/or number if
2754 the presentation setting permits it.
2755 * Added display_setup parameter to misdn.conf to put a display string
2756 in the SETUP message containing the caller name and/or number if the
2757 presentation setting permits it.
2758 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2759 indicate the dialplan settings are to be obtained from the asterisk
2761 * Made misdn.conf parameter callerid accept the "name" <number> format
2762 used by the rest of the system.
2763 * Made use the nationalprefix and internationalprefix misdn.conf
2764 parameters to prefix any received number from the ISDN link if that
2765 number has the corresponding Type-Of-Number. NOTE: This includes
2766 comparing the incoming call's dialed number against the MSN list.
2767 * Added the following new parameters: unknownprefix, netspecificprefix,
2768 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2769 received number from the ISDN link if that number has the corresponding
2771 * Added new dialplan application misdn_command which permits controlling
2772 the CCBS/CCNR functionality.
2773 * Added new dialplan function mISDN_CC which permits retrieval of various
2774 values from an active call completion record.
2775 * For PTP, you should manually send the COLR of the redirected-to party
2776 for an incomming redirected call if the incoming call could experience
2777 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2778 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2779 if the REDIRECTING(from-num) is not empty.
2780 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2781 option on all of the REDIRECTING statements before dialing the
2782 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2783 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2784 redirecting-to presentation (COLR) when it becomes available.
2785 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2788 thirdparty mISDN enhancements
2789 -----------------------------
2790 mISDN has been modified by Digium, Inc. to greatly expand facility message
2792 * Enhanced COLP support for call diversion and transfer.
2793 * CCBS/CCNR support.
2795 The latest modified mISDN v1.1.x based version is available at:
2796 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2797 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2799 Tagged versions of the modified mISDN code are available under:
2800 http://svn.digium.com/svn/thirdparty/mISDN/tags
2801 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2803 libpri channel driver (chan_dahdi) DAHDI changes
2804 -------------------------------------------
2805 * The channel variable PRIREDIRECTREASON is now just a status variable
2806 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2807 to read and alter the reason.
2808 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2809 redirected-to party for an incomming redirected call if the incoming call
2810 could experience further redirects. Just set the
2811 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2812 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2814 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2815 use the inhibit(i) option on all of the REDIRECTING statements before
2816 dialing the redirected-to party. You still have to set the
2817 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2818 will update the redirecting-to presentation (COLR) when it becomes available.
2819 * Added the ability to ignore calls that are not in a Multiple Subscriber
2820 Number (MSN) list for PTMP CPE interfaces.
2821 * Added dynamic range compression support for dahdi channels. It is
2822 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2823 * Added support for ISDN calling and called subaddress with partial support
2824 for connected line subaddress.
2825 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2826 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2827 to transfer a held call on disconnect similar to an analog phone.
2828 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2829 Will reroute/deflect an outgoing call when receive the message.
2830 Can use the DAHDISendCallreroutingFacility to send the message for the
2832 * Added standard location to add options to chan_dahdi dialing:
2833 Dial(DAHDI/g1[/extension[/options]])
2836 R Reverse charging indication
2837 * Added Reverse Charging Indication (Collect calls) send/receive option.
2838 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2839 Dial(DAHDI/g1/extension/R)
2840 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2841 (requires latest LibPRI)
2842 * Added ability to send/receive keypad digits in the SETUP message.
2843 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2844 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2845 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2846 (requires latest LibPRI)
2847 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2848 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2849 back into the same interface. Tromboned calls happen because of call routing,
2850 call deflection, call forwarding, and call transfer.
2851 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2852 * Added the ability to support call waiting calls. (The SETUP has no B channel
2854 * Added Malicious Call ID (MCID) event to the AMI call event class.
2855 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2857 Asterisk Manager Interface
2858 --------------------------
2859 * The Hangup action now accepts a Cause header which may be used to
2860 set the channel's hangup cause.
2861 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2862 to specify a separate .pem file to hold a private key. By default sslcert
2863 is used to hold both the public and private key.
2864 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2865 for options containing the 'tls' prefix. For example, 'sslenable' is now
2866 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2867 across all .conf files. All affected sample.conf files have been modified to
2868 reflect this change. Previous options such as 'sslenable' still work,
2869 but options with the 'tls' prefix are preferred.
2870 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2871 in a channel. (res_mutestream.so)
2872 * The configuration file manager.conf now supports a channelvars option, which
2873 specifies a list of channel variables to include in each channel-oriented
2875 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2876 and ExtraPriority to allow redirecting the second channel to a different
2877 location than the first.
2878 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2880 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2881 in a MixMonitor recording.
2882 * The 'iax2 show peers' output is now similar to the expected output of
2884 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2886 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2887 AOC-E messages on a channel.
2888 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2889 conform more closely to similar events.
2890 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2892 * Added optional parkinglot variable for park command.
2893 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2894 if CallerIDNum and CallerIDName headers are also present.
2896 Channel Event Logging
2897 ---------------------
2898 * A new interface, CEL, is introduced here. CEL logs single events, much like
2899 the AMI, but it differs from the AMI in that it logs to db backends much
2900 like CDR does; is based on the event subsystem introduced by Russell, and
2901 can share in all its benefits; allows multiple backends to operate like CDR;
2902 is specialized to event data that would be of concern to billing sytems,
2903 like CDR. Backends for logging and accounting calls have been produced,
2904 but a new CDR backend is still in development.
2908 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2909 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2910 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2911 * Multiple files and formats can now be specified in cdr_custom.conf.
2912 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2913 See configs/cdr_syslog.conf.sample for more information.
2914 * A 'sequence' field has been added to CDRs which can be combined with
2915 linkedid or uniqueid to uniquely identify a CDR.
2916 * Handling of billsec and duration field has changed. If your table definition
2917 specifies those fields as float,double or similar they will now be logged with
2918 microsecond accuracy instead of a whole integer.
2920 Calendaring for Asterisk
2921 ------------------------
2922 * A new set of modules were added supporing calendar integration with Asterisk.
2923 Dialplan functions for reading from and writing to calendars are included,
2924 as well as the ability to execute dialplan logic upon calendar event notifications.
2925 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2926 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2927 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2928 2003 support does not support forms-based authentication).
2930 Call Completion Supplementary Services for Asterisk
2931 ---------------------------------------------------
2932 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2933 DAHDI/ISDN supports call completion for the following switch types:
2934 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2935 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2937 Multicast RTP Support
2938 ---------------------
2939 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2940 The channel driver can be used with the Page application to perform multicast RTP
2941 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2942 Type can be either basic or linksys.
2943 Destination is the IP address and port for the RTP packets.
2944 Control address is specific to the linksys type and is used for sending the control
2945 packets unique to them.
2947 Security Events Framework
2948 -------------------------
2949 * Asterisk has a new C API for reporting security events. The module res_security_log
2950 sends these events to the "security" logger level. Currently, AMI is the only
2951 Asterisk component that reports security events. However, SIP support will be
2952 coming soon. For more information on the security events framework, see the
2953 "Asterisk Security Framework" section of the Asterisk wiki at
2954 https://wiki.asterisk.org/wiki/x/wgBQ
2955 * SIP support was added in Asterisk 10
2956 * This API now supports IPv6 addresses
2960 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2961 * A spandsp based fax backend (res_fax_spandsp) has been added.
2962 * The app_fax module has been deprecated in favor of the res_fax module and
2963 the new res_fax_spandsp backend.
2964 * The SendFAX and ReceiveFAX applications now send their log messages to a
2965 'fax' logger level, instead of to the generic logger levels. To see these
2966 messages, the system's logger.conf file will need to direct the 'fax' logger
2967 level to one or more destinations; the logger.conf.sample file includes an
2968 example of how to do this. Note that if the 'fax' logger level is *not*
2969 directed to at least one destination, log messages generated by these
2970 applications will be lost, and that if the 'fax' logger level is directed to
2971 the console, the 'core set verbose' and 'core set debug' CLI commands will
2972 have no effect on whether the messages appear on the console or not.
2976 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2977 Now, in order to enable transmitting silence during record the transmit_silence
2978 option should be used. transmit_silence_during_record remains a valid option, but
2979 defaults to the behavior of the transmit_silence option.
2980 * Addition of the Unit Test Framework API for managing registration and execution
2981 of unit tests with the purpose of verifying the operation of C functions.
2982 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2983 XMPP text messages to the remote JID.
2984 * Modules.conf has a new option - "require" - that marks a module as critical for
2985 the execution of Asterisk.
2986 If one of the required modules fail to load, Asterisk will exit with a return
2988 * An 'X' option has been added to the asterisk application which enables #exec support.
2989 This allows #exec to be used in asterisk.conf.
2990 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2991 * A new lockconfdir option has been added to asterisk.conf to protect the
2992 configuration directory (/etc/asterisk by default) during reloads.
2993 * The parkeddynamic option has been added to features.conf to enable the creation
2994 of dynamic parkinglots.
2995 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2996 the reportalarms config option.
2997 * chan_dahdi supports dialing configuring and dialing by device file name.
2998 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2999 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3000 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3001 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3002 Handy for the above name-based syntax as it does not depend on
3003 initialization order.
3004 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3005 significant increase in performance (about 3X) for installations using this switchtype.
3006 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3007 AIS. For more information, please see the Distributed Device State section of the
3008 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3009 * The addition of G.719 pass-through support.
3010 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3011 during device configuration.
3012 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3013 have less than 3 lines on the LCD.
3014 * Realtime now supports database failover. See the sample extconfig.conf for details.
3015 * The addition of improved translation path building for wideband codecs. Sample
3016 rate changes during translation are now avoided unless absolutely necessary.
3017 * The addition of the res_stun_monitor module for monitoring and reacting to network
3018 changes while behind a NAT.
3019 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3020 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3021 These allow support for any Administration. Default is AT&T values.
3025 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3026 optionally accept a filename, to apply the setting only to the code generated from
3027 that source file when Asterisk was built. However, there are some modules in Asterisk
3028 that are composed of multiple source files, so this did not result in the behavior
3029 that users expected. In this version, 'core set debug' and 'core set verbose'
3030 can optionally accept *module* names instead (with or without the .so extension),
3031 which applies the setting to the entire module specified, regardless of which source
3032 files it was built from.
3033 * New 'manager show settings' command showing the current settings loaded from
3035 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3036 the channel hangup request to all channels.
3037 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3039 ------------------------------------------------------------------------------
3040 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3041 ------------------------------------------------------------------------------
3045 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3046 Snom phones use this for call pickup of extensions that the phone is
3048 * Added support for setting the domain in the URI for caller of an
3049 outbound call by using the SIPFROMDOMAIN channel variable.
3050 * Added a new configuration option "remotesecret" for authentication to
3051 remote services. For backwards compatibility, "secret" still has the
3052 same function as before, but now you can configure both a remote secret and a
3053 local secret for mutual authentication.
3054 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3055 the sound will be played to the target of an attended transfer
3056 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3057 finer control over how many peers Asterisk will qualify and the gap between them
3058 when all peers need to be qualified at the same time.
3059 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3060 (either globally or for a specific peer), chan_sip will treat any SDP data
3061 it receives as new data and update the media stream accordingly. By
3062 default, Asterisk will only modify the media stream if the SDP session
3063 version received is different from the current SDP session version. This
3064 option is required to interoperate with devices that have non-standard SDP
3065 session version implementations (observed with Microsoft OCS). This option
3066 is disabled by default.
3067 * The parsing of register => lines in sip.conf has been modified to allow a port
3068 to be present in the "user" portion. Please see the sip.conf.sample file for more
3070 * Added support for subscribing to MWI on a remote server and making the status available
3071 as a mailbox. Please see the sip.conf.sample file for more information.
3072 * Added a function to remove SIP headers added in the dialplan before the
3073 first INVITE is generated - SIPRemoveHeader()
3074 * Channel variables set with setvar= in a device configuration is now
3075 set both for inbound and outbound calls.
3076 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3080 * Added immediate option to iax.conf
3081 * Added forceencryption option to iax.conf
3082 * Added Encryption and Trunk status to manager command "iaxpeers"
3086 * The configuration file now holds separate sections for devices and lines.
3087 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3092 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3093 support for LibOpenR2. http://www.libopenr2.org/
3094 * The UK option waitfordialtone has been added for use with BT analog
3096 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3097 is used in conjunction with the 'faxdetect' configuration option. When
3098 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3099 switch to the configured faxbuffers policy. For example, to use 6 buffers
3100 and a 'full' buffer policy for a fax transmission, add:
3102 The faxbuffers configuration will be in affect until the call is torn down.
3103 * Added service message support for 4ESS/5ESS switches.
3107 * For DAHDI channels, the CHANNEL() dialplan function now
3108 supports changing the channel's buffer policy (for the current
3109 call only), using this syntax:
3111 exten => s,n,Set(CHANNEL(buffers)=6,full)
3113 This would change the channel to the 'full' buffer policy and
3114 6 (six) buffers. Possible options for this setting are the same
3115 as those in chan_dahdi.conf.
3116 * Added a new dialplan function, CURLOPT, which permits setting various
3117 options that may be useful with the CURL dialplan function, such as
3118 cookies, proxies, connection timeouts, passwords, etc.
3119 * Permit the syntax and synopsis fields of the corresponding dialplan
3120 functions to be individually set from func_odbc.conf.
3121 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3122 * func_odbc now may specify an insert query to execute, when the write query
3123 affects 0 rows (usually indicating that no such row exists).
3124 * Added a new dialplan function, LISTFILTER, which permits removing elements
3125 from a set list, by name. Uses the same general syntax as the existing CUT
3126 and FIELDQTY dialplan functions, which also manage lists.
3127 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3128 obtaining realtime data from the dialplan.
3129 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3130 a subroutine when using the GoSub() and Return() applications.
3131 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3132 of "core show function AUDIOHOOK_INHERIT" from the CLI
3133 * Added AES_ENCRYPT. For information on its use, please see the output
3134 of "core show function AES_ENCRYPT" from the CLI
3135 * Added AES_DECRYPT. For information on its use, please see the output
3136 of "core show function AES_DECRYPT" from the CLI
3137 * func_odbc now supports database transactions across multiple queries.
3141 * Scheduled meetme conferences may now have their end times extended by
3143 * app_authenticate now gives the ability to select a prompt other than
3145 * app_directory now pays attention to the searchcontexts setting in
3146 voicemail.conf and will look through all contexts, if no context is
3147 specified in the initial argument.
3148 * A new application, Originate, has been introduced, that allows asynchronous
3149 call origination from the dialplan.
3150 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3151 in addition to the setting in the "general" context.
3152 * Added ConfBridge dialplan application which does conference bridges without
3153 DAHDI. For information on its use, please see the output of
3154 "core show application ConfBridge" from the CLI.
3158 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3159 operation to the AMI Redirect action.
3160 * extensions.conf now allows you to use keyword "same" to define an extension
3161 without actually specifying an extension. It uses exactly the same pattern
3162 as previously used on the last "exten" line. For example:
3163 exten => 123,1,NoOp(something)
3164 same => n,SomethingElse()
3165 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3166 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3167 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3168 by the new clialiases module. See cli_aliases.conf.sample file.
3169 * Times within timespecs are now accurate down to the minute. This is a change
3170 from historical Asterisk, which only provided timespecs rounded to the nearest
3171 even (read: evenly divisible by 2) minute mark.
3172 * The realtime switch now supports an option flag, 'p', which disables searches for
3174 * In addition to a time range and date range, timespecs now accept a 5th optional
3175 argument, timezone. This allows you to perform time checks on alternate
3176 timezones, especially if those daylight savings time ranges vary from your
3177 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3179 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3180 give you the correct output for an asterisk box behind nat. It will give you the
3181 externhost and localnet settings.
3182 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3183 can connect calls in passthrough mode, as well as record and play back files.
3184 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3185 using pickupsound and pickupfailsound in features.conf.
3186 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3187 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3188 instead of the /var/run/asterisk.pid where it used to be. This will make
3189 installs as non-root easier to manage.
3194 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3195 be written; they will no longer be explicitly written.
3197 Asterisk Manager Interface
3198 --------------------------
3199 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3200 a non-empty value) in your request. If you do this, any pending AMI events will
3201 *not* be included in the response to your request as they would normally, but
3202 will be left in the event queue for the next request you make to retrieve. For
3203 some applications, this will allow you to guarantee that you will only see
3204 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3205 To know whether the Asterisk server supports this header or not, your client can
3206 inspect the first response back from the server to see if it includes this header:
3208 Pragma: SuppressEvents
3210 If this is included, the server supports event suppression.
3212 * Added 4 new Actions to list skinny device(s) and line(s)
3218 LDAP Schema File Additions
3219 --------------------------
3220 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3221 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3223 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3224 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3225 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3226 * Removed redundant IPaddr (there's already IPAddress)
3227 - Gives more configuration Flags for SIP-Users available (tested)
3228 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3229 without extensibleObject (which really should be the last resort); gives
3230 also additional possibilities for LDAP-filter
3232 ------------------------------------------------------------------------------
3233 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3234 ------------------------------------------------------------------------------
3236 Device State Handling
3237 ---------------------
3238 * The event infrastructure in Asterisk got another big update to help support
3239 distributed events. It currently supports distributed device state and
3240 distributed Voicemail MWI (Message Waiting Indication). A new module has
3241 been merged, res_ais, which facilitates communicating events between servers.
3242 It uses the SAForum AIS (Service Availability Forum Application Interface
3243 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3244 a cluster of Asterisk servers, and to share events between them. For more
3245 information on setting this up, refer to the Distributed Device State section
3246 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3250 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3251 variables from an Asterisk configuration file.
3252 * The JACK_HOOK function now has a c() option to supply a custom client name.
3253 * Added two new dialplan functions from libspeex for audio gain control and
3254 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3255 rx directions of a channel from the dialplan.
3256 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3257 based on other parameters. The default is still to search based on the
3258 forwarding station ID. However, there are new options that allow you to search
3259 based on the message desk terminal ID, or the message desk number.
3260 * TIMEOUT() has been modified to be accurate down to the millisecond.
3261 * ENUM*() functions now include the following new options:
3262 - 'u' returns the full URI and does not strip off the URI-scheme.
3263 - 's' triggers ISN specific rewriting
3264 - 'i' looks for branches into an Infrastructure ENUM tree
3265 - 'd' for a direct DNS lookup without any flipping of digits.
3266 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3267 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3268 deviation of jitter, rtt, and loss for a call using chan_sip.
3270 DAHDI channel driver (chan_dahdi) Changes
3271 ----------------------------------------
3272 * Channels can now be configured using named sections in chan_dahdi.conf, just
3273 like other channel drivers, including the use of templates.
3274 * The default for pridialplan has changed from 'national' to 'unknown'.
3278 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3279 to something that matches the pattern a hint will be created using the contents
3280 and variables evaluated.
3281 * Dialplan matching has been extended to allow an extension to return to the
3282 PBX core to wait for more digits. This is done by using the new dialplan
3283 application called "Incomplete". This will permit a whole new level of
3284 extension control, by giving the administrator more control over early
3285 matches employing one of the short-circuit pattern match operators. Note
3286 that custom applications can trigger this same behavior by returning the
3287 special value AST_PBX_INCOMPLETE.
3291 * Directory now permits both first and last names to be matched at the same
3292 time. In addition, the number of digits to enter of the name can be set in
3293 the arguments to Directory; previously, you could enter only 3, regardless
3294 of how many names are in your company. For large companies, this should be
3296 * Voicemail now permits a mailbox setting to wrap around from first to last
3297 messages, if the "messagewrap" option is set to a true value.
3298 * Voicemail now permits an external script to be run, for password validation.
3299 The script should output "VALID" or "INVALID" on stdout, depending upon the
3300 wish to validate or invalidate the password given. Arguments are:
3301 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3303 * Dial has a new option: F(context^extension^pri), which permits a callee to
3304 continue in the dialplan, at the specified label, if the caller hangs up.
3305 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3306 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3307 * The Jack application now has a c() option to supply a custom client name.
3308 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3309 like the pre-existing whisper mode, except that the spy can also talk to the
3310 participant on the bridged channel as well.
3311 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3312 to be spoken instead of the channel name or number. For more information on the
3313 use of this option, issue the command "core show application ChanSpy" from the
3315 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3316 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3317 words, if using the 'd' option, it is not possible to enter a number to append to
3318 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3319 change to whisper mode, and pressing 6 will change to barge mode.
3320 * ExternalIVR now takes several options that affect the way it performs, as
3321 well as having several new commands. Please see the External IVR page on the Asterisk
3322 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3323 * Added ability to communicate over a TCP socket instead of forking a child process for the
3324 ExternalIVR application.
3325 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3326 of just the first one if you give the function more then one channel to check.
3327 * PrivacyManager now takes an option where you can specify a context where the
3328 given number will be matched. This way you have more control over who is allowed
3329 and it stops the people who blindly enter 10 digits.
3330 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3331 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3332 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3333 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3334 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3335 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3336 * The Dial() application no longer copies the language used by the caller to the callee's
3337 channel. If you desire for the caller's channel's language to be used for file playback
3338 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3339 * SendImage() no longer hangs up the channel on error; instead, it sets the
3340 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3341 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3343 * Park has a new option, 's', which silences the announcement of the parking space number.
3344 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3345 invalid input and will be assumed to mean that no timeout is desired.
3349 * Added DNS manager support to registrations for peers referencing peer entries.
3350 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3351 as well as periodically updating the IP address. These properties allow for
3352 better performance as well as recovery in the event of an IP change.
3353 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3354 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3355 These changes also provide performance improvements for call setup and tear down.
3356 * Added ability to specify registration expiry time on a per registration basis in
3358 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3360 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3361 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3362 * 'sip show peers' and 'sip show users' display their entries sorted in
3363 alphabetical order, as opposed to the order they were in, in the config
3365 * Videosupport now supports an additional option, "always", which always sets
3366 up video RTP ports, even on clients that don't support it. This helps with
3367 callfiles and certain transfers to ensure that if two video phones are
3368 connected, they will always share video feeds.
3372 * Existing DNS manager lookups extended to check for SRV records.
3373 * IAX2 encryption support has been improved to support periodic key rotation
3374 within a call for enhanced security. The option "keyrotate" has been
3375 provided to disable this functionality to preserve backwards compatibility
3376 with older versions of IAX2 that do not support key rotation.
3380 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3381 data tree based on the given <path>.
3382 * New CLI command "data show providers" that will display all the registered
3384 * New CLI command, "config reload <file.conf>" which reloads any module that
3385 references that particular configuration file. Also added "config list"
3386 which shows which configuration files are in use.
3387 * New CLI commands, "pri show version" and "ss7 show version" that will
3388 display which version of libpri and libss7 are being used, respectively.
3389 A new API call was added so trunk will now have to be compiled against
3390 a versions of libpri and libss7 that have them or it will not know that
3391 these libraries exist.
3392 * The commands "core show globals", "core set global" and "core set chanvar" has
3393 been deprecated in favor of the more semanticly correct "dialplan show globals",
3394 "dialplan set chanvar" and "dialplan set global".
3395 * New CLI command "dialplan show chanvar" to list all variables associated
3396 with a given channel.
3400 * Addresses managed by DNS manager now can check to see if there is a DNS
3401 SRV record for a given domain and will use that hostname/port if present.
3403 AMI - The manager (TCP/TLS/HTTP)
3404 --------------------------------
3405 * The Status command now takes an optional list of variables to display
3406 along with channel status.
3407 * The QueueEntry event now also includes the channel's uniqueid
3411 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3412 as some people were running into this limit. This limit has been increased
3417 * The TRANSFER queue log entry now includes the the caller's original
3418 position in the transferred-from queue.
3419 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3420 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3421 as well as an explanation about timeout options in general
3422 * Added a new option - C - for forcing the "answered elsewhere" flag on
3423 cancellation of calls in to members of the queue. This is to avoid the
3424 call to a member of a queue having the call listed as a "missed call".
3428 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3429 adaptive capabilities. What this means in practical terms is that if your
3430 realtime table lacks critical fields, Asterisk will now emit warnings to
3431 that effect. Also, some of the realtime drivers have the ability (if
3432 configured) to automatically add those columns to the table with the
3433 correct type and length.
3437 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3438 the 'setvar' option to cause a given audio file to be played upon completion
3439 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3440 Skinny channels only.
3441 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3442 Hoard page on the Asterisk wiki for more information:
3443 https://wiki.asterisk.org/wiki/x/pQBB
3444 * Config file variables may now be appended to, by using the '+=' append
3445 operator. This is most helpful when working with long SQL queries in
3446 func_odbc.conf, as the queries no longer need to be specified on a single
3448 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3449 which will add a second to the billsec when the ending
3450 time is set, if the number in the microseconds field of the end time is
3451 greater than the number of microseconds in the answer time. This allows
3452 users to count the 'initiated' seconds in their billing records.
3454 ------------------------------------------------------------------------------
3455 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3456 ------------------------------------------------------------------------------
3458 AMI - The manager (TCP/TLS/HTTP)
3459 --------------------------------
3460 * Manager has undergone a lot of changes, all of them documented
3461 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3462 * Manager version has changed to 1.1
3463 * Added a new action 'CoreShowChannels' to list currently defined channels
3464 and some information about them.
3465 * Added a new action 'SIPshowregistry' to list SIP registrations.
3466 * Added TLS support for the manager interface and HTTP server
3467 * Added the URI redirect option for the built-in HTTP server
3468 * The output of CallerID in Manager events is now more consistent.
3469 CallerIDNum is used for number and CallerIDName for name.
3470 * Enable https support for builtin web server.
3471 See configs/http.conf.sample for details.
3472 * Added a new action, GetConfigJSON, which can return the contents of an
3473 Asterisk configuration file in JSON format. This is intended to help
3474 improve the performance of AJAX applications using the manager interface
3476 * SIP and IAX manager events now
projects / asterisk/asterisk.git / blob