1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
13 ------------------------------------------------------------------------------
16 --------------------------
19 --------------------------
20 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
21 as the chanprefix parameter if the 'u' option is specified.
24 --------------------------
25 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
26 conference user menus.
28 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
29 menus, bridge settings, and user settings that have been applied by the
30 CONFBRIDGE dialplan function.
32 * The ConfBridge dialplan application now sets a channel variable,
33 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
34 how a channel exited the conference.
36 * Added conference user option 'announce_join_leave_review'. This option
37 implies 'announce_join_leave' with the added effect that the user will
38 be asked if they want to confirm or re-record the recording of their
39 name when entering the conference
42 --------------------------
43 * At exit, the Directory application now sets a channel variable
44 DIRECTORY_RESULT to one of the following based on the reason for exiting:
45 OPERATOR user requested operator by pressing '0' for operator
46 ASSISTANT user requested assistant by pressing '*' for assistant
47 TIMEOUT user pressed nothing and Directory stopped waiting
48 HANGUP user's channel hung up
49 SELECTED user selected a user from the directory and is routed
50 USEREXIT user pressed '#' from the selection prompt to exit
51 FAILED directory failed in a way that wasn't accounted for. Dang.
54 --------------------------
55 * MusicOnHold streams (all modes other than "files") now support wide band
59 --------------------------
60 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
61 and for the channel executing Page respectively.
64 --------------------------
65 * PickupChan now accepts channel uniqueids of channels to pickup.
68 --------------------------
69 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
70 to 'true' (case insensitive), then any Say application (SayNumber,
71 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
72 anticipate DTMF. If DTMF is received, these applications will behave like
73 the background application and jump to the received extension once a match
74 is established or after a short period of inactivity.
77 -------------------------
78 * A new function, MIXMONITOR, has been added to allow access to individual
79 instances of MixMonitor on a channel.
82 -------------------------
83 * Core Show Locks output now includes Thread/LWP ID if the platform
84 supports this feature.
86 ------------------------------------------------------------------------------
87 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
88 ------------------------------------------------------------------------------
92 * Added a new module that provides AMI control over MWI within Asterisk,
93 res_mwi_external_ami. Note that this module depends on res_mwi_external;
94 for more information on enabling this module, see res_mwi_external.
95 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
96 the MWIGet/MWIGetComplete events.
98 * The DialStatus field in the DialEnd event can now contain additional
99 statuses that convey how the dial operation terminated. This includes
100 ABORT, CONTINUE, and GOTO.
104 * The Bridge object contains new fields 'name' and 'creator'. The name
105 is a special description for the bridge given to it upon creation. The
106 creator is the name of the service/module/etc that created the bridge.
108 * Added a new ARI resource 'mailboxes' which allows the creation and
109 modification of mailboxes managed by external MWI. Modules res_mwi_external
110 and res_stasis_mailbox must be enabled to use this resource. For more
111 information on external MWI control, see res_mwi_external.
113 * Added new events for externally initiated transfers. The event
114 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
115 of a bridge in the ARI controlled application to the dialplan; the
116 BridgeAttendedTransfer event is raised when a channel initiates an
117 attended transfer of a bridge in the ARI controlled application to the
120 * Channel variables may now be specified as a body parameter to the
121 POST /channels operation. The 'variables' key in the JSON is interpreted
122 as a sequence of key/value pairs that will be added to the created channel
123 as channel variables. Other parameters in the JSON body are treated as
124 query parameters of the same name.
128 * Path support has been added with the 'support_path' option in registration
131 * A 'debug' option has been added to the globals section that will allow
132 sip messages to be logged.
134 * A 'set_var' option has been added to endpoints that will automatically
135 set the desired variable(s) on a channel created for that endpoint.
137 * Several new tables and columns have been added to the realtime schema for
138 the res_pjsip related modules. See the UPGRADE.txt notes for updating
143 * A new module, res_mwi_external, has been added to Asterisk. This module
144 acts as a base framework that other modules can build on top of to allow
145 an external system to control MWI within Asterisk. For implementations
146 that make use of res_mwi_external, see res_mwi_external_ami and
147 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
148 that may produce MWI themselves, such as app_voicemail. res_mwi_external
149 and other modules that depend on it cannot be built or loaded with
150 app_voicemail present.
153 ------------------------------------------------------------------------------
154 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
155 ------------------------------------------------------------------------------
160 Asterisk 12 is a standard release of the Asterisk project. As such, the
161 focus of development for this release was on core architectural changes and
162 major new features. This includes:
163 * A more flexible bridging core based on the Bridging API
164 * A new internal message bus, Stasis
165 * Major standardization and consistency improvements to AMI
166 * Addition of the Asterisk RESTful Interface (ARI)
167 * A new SIP channel driver, chan_pjsip
168 In addition, as the vast majority of bridging in Asterisk was migrated to the
169 Bridging API used by ConfBridge, major changes were made to most of the
170 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
172 Specifications have been written for the affected interfaces. These
173 specifications are available on the Asterisk wiki:
174 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
175 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
176 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
178 It is *highly* recommended that anyone migrating to Asterisk 12 read the
179 information regarding its release both in this file and in the accompanying
180 UPGRADE.txt file. More detailed information on the major changes can be found
181 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
186 * Added build option DISABLE_INLINE. This option can be used to work around a
187 bug in gcc. For more information, see
188 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
190 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
191 the CHANNEL_TRACE build option were incompatible with the new bridging
194 * Asterisk now optionally uses libxslt to improve XML documentation generation
195 and maintainability. If libxslt is not available on the system, some XML
196 documentation will be incomplete.
198 * Asterisk now depends on libjansson. If a package of libjansson is not
199 available on your distro, please see http://www.digip.org/jansson/.
201 * Asterisk now depends on libuuid and, optionally, uriparser. It is
202 recommended that you install uriparser, even if it is optional.
204 * The new SIP stack and channel driver uses a particular version of PJSIP.
205 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
206 configuring and installing PJSIP for usage with Asterisk.
208 * Optional API was re-implemented to be more portable, and no longer requires
209 weak reference support from the compiler. The build option OPTIONAL_API may
210 be disabled to disable Optional API support.
217 * Along with AgentRequest, this application has been modified to be a
218 replacement for chan_agent. The act of a channel calling the AgentLogin
219 application places the channel into a pool of agents that can be
220 requested by the AgentRequest application. Note that this application, as
221 well as all other agent related functionality, is now provided by the
222 app_agent_pool module. See chan_agent and AgentRequest for more information.
224 * This application no longer performs agent authentication. If authentication
225 is desired, the dialplan needs to perform this function using the
226 Authenticate or VMAuthenticate application or through an AGI script before
229 * If this application is called and the agent is already logged in, the
230 dialplan will continue exection with the AGENT_STATUS channel variable set
231 to ALREADY_LOGGED_IN.
233 * The agents.conf schema has changed. Rather than specifying agents on a
234 single line in comma delineated fashion, each agent is defined in a separate
235 context. This allows agents to use the power of context templates in their
238 * A number of parameters from agents.conf have been removed. This includes
239 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
240 urlprefix, and savecallsin. These options were obsoleted by the move from
241 a channel driver model to the bridging/application model provided by
246 * A new application, this will request a logged in agent from the pool and
247 bridge the requested channel with the channel calling this application.
248 Logged in agents are those channels that called the AgentLogin application.
249 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
250 application will be set with an appropriate error value.
254 * This application has been removed. It was a holdover from when
255 AgentCallbackLogin was removed.
259 * Added support for additional Ademco DTMF signalling formats, including
260 Express 4+1, Express 4+2, High Speed and Super Fast.
262 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
263 call time, in milliseconds, to run the application.
265 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
266 maximum number of times to retry the call.
268 * Added a new configuration option answait. If set, the AlarmReceiver
269 application will wait the number of milliseconds specified by answait
270 after the channel has answered. Valid values range between 500
271 milliseconds and 10000 milliseconds.
273 * Added configuration option no_group_meta. If enabled, grouping of metadata
274 information in the AlarmReceiver log file will be skipped.
278 * It is now no longer possible to bypass updating the CDR on the channel
279 when answering. CDRs reflect the state of the channel and will always
280 reflect the time they were Answered.
284 * A new application in Asterisk, this will place the calling channel
285 into a holding bridge, optionally entertaining them with some form of
286 media. Channels participating in a holding bridge do not interact with
287 other channels in the same holding bridge. Optionally, however, a channel
288 may join as an announcer. Any media passed from an announcer channel is
289 played to all channels in the holding bridge. Channels leave a holding
290 bridge either when an optional timer expires, or via the ChannelRedirect
291 application or AMI Redirect action.
295 * All participants in a bridge can now be kicked out of a conference room
296 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
297 command, i.e., 'confbridge kick <conference> all'
299 * CLI output for the 'confbridge list' command has been improved. When
300 displaying information about a particular bridge, flags will now be shown
301 for the participating users indicating properties of that user.
303 * The ConfbridgeList event now contains the following fields: WaitMarked,
304 EndMarked, and Waiting. This displays additional properties about the
305 user's profile, as well as whether or not the user is waiting for a
306 Marked user to enter the conference.
308 * Added a new option for conference recording, record_file_append. If enabled,
309 when the recording is stopped and then re-started, the existing recording
310 will be used and appended to.
312 * ConfBridge now has the ability to set the language of announcements to the
313 conference. The language can be set on a bridge profile in confbridge.conf
314 or by the dialplan function CONFBRIDGE(bridge,language)=en.
318 * The channel variable CPLAYBACKSTATUS may now return the value
319 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
320 such as AMI. See the AMI action ControlPlayback for more information.
324 * Added the 'a' option, which allows the caller to enter in an additional
325 alias for the user in the directory. This option must be used in conjunction
326 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
327 specified in voicemail.conf.
331 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
332 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
333 containing the unique ID of the bridge that the channel happens to be in.
337 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
338 for more information.
340 * Variables are no longer purged from the original CDR. See the 'v' option for
343 * The 'A' option has been removed. The Answer time on a CDR is never updated
346 * The 'd' option has been removed. The disposition on a CDR is a function of
347 the state of the channel and cannot be altered.
349 * The 'D' option has been removed. Who the Party B is on a CDR is a function
350 of the state of the respective channels involved in the CDR and cannot be
353 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
354 such that the start time and, if applicable, the answer time was updated.
355 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
356 'r' option now triggers the Reset, setting the start time (and answer time
357 if applicable) to the current time. Note that the 'a' option still sets
358 the answer time to the current time if the channel was already answered.
360 * The 's' option has been removed. A variable can be set on the original CDR
361 if desired using the CDR function, and removed from a forked CDR using the
364 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
365 longer applies in the CDR engine.
367 * The 'v' option now prevents the copy of the variables from the original CDR
368 to the forked CDR. Previously the variables were always copied but were
369 removed from the original. This was changed as removing variables from a CDR
370 can have unintended side effects - this option allows the user to prevent
371 propagation of variables from the original to the forked without modifying
376 * Added the 'n' option to MeetMe to prevent application of the DENOISE
377 function to a channel joining a conference. Some channel drivers that vary
378 the number of audio samples in a voice frame will experience significant
379 quality problems if a denoiser is attached to the channel; this option gives
380 them the ability to remove the denoiser without having to unload func_speex.
384 * The 'b' option now includes conferences as well as sounds played to the
387 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
388 running during a transfer. If a MixMonitor is started on a channel,
389 the MixMonitor will continue to record the audio passing through the
390 channel even in the presence of transfers.
394 * The NoCDR application is deprecated. Please use the CDR_PROP function to
397 * While the NoCDR application will prevent CDRs for a channel from being
398 propagated to registered CDR backends, it will not prevent that data from
399 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
400 function that enables CDRs on a channel will restore those records that have
401 not yet been finalized.
405 * The app_parkandannounce module has been removed. The application
406 ParkAndAnnounce is now provided by the res_parking module. See the
407 res_parking changes for more information.
411 * Added queue available hint. The hint can be added to the dialplan using the
412 following syntax: exten,hint,Queue:{queue_name}_avail
413 For example, if the name of the queue is 'markq':
414 exten => 8501,hint,Queue:markq_avail
415 This will report 'InUse' if there are no logged in agents or no free agents.
416 It will report 'Idle' when an agent is free.
418 * Queues now support a hint for member paused state. The hint uses the form
419 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
420 are the name of the queue and the name of the member to subscribe to,
421 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
422 Members will show as In Use when paused.
424 * The configuration options eventwhencalled and eventmemberstatus have been
425 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
426 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
427 sent. The "Variable" fields will also no longer exist on the Agent* events.
428 These events can be filtered out from a connected AMI client using the
429 eventfilter setting in manager.conf.
431 * The queue log now differentiates between blind and attended transfers. A
432 blind transfer will result in a BLINDTRANSFER message with the destination
433 context and extension. An attended transfer will result in an
434 ATTENDEDTRANSFER message. This message will indicate the method by which
435 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
436 for running an application on a bridge or channel, or "LINK" for linking
437 two bridges together with local channels. The queue log will also now detect
438 externally initiated blind and attended transfers and record the transfer
441 * When performing queue pause/unpause on an interface without specifying an
442 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
443 least one member of any queue exists for that interface.
445 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
446 for realtime queue log entries.
450 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
451 CDRs when they were previously disabled on a channel.
453 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
454 backends occurs on an as-needed basis in order to preserve linkedid
455 propagation and other needed behavior.
459 * A new application, this is similar to SayAlpha except that it supports
460 case sensitive playback of the specified characters. For example,
461 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
465 * This application is deprecated in favor of CHANNEL(amaflags).
469 * The SendDTMF application will now accept 'W' as valid input. This will cause
470 the application to delay one second while streaming DTMF.
474 * A new application in Asterisk 12, this hands control of the channel calling
475 the application over to an external system. Currently, external systems
476 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
480 * UserEvent will now handle duplicate keys by overwriting the previous value
483 * In addition to AMI, UserEvent invocations will now be distributed to any
484 interested Stasis applications.
488 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
489 system as mailbox@context. The rest of the system cannot add @default
490 to mailbox identifiers for app_voicemail that do not specify a context
491 any longer. It is a mailbox identifier format that should only be
492 interpreted by app_voicemail.
494 * The voicemail.conf configuration file now has an 'alias' configuration
495 parameter for use with the Directory application. The voicemail realtime
496 database table schema has also been updated with an 'alias' column.
501 * Pass through support has been added for both VP8 and Opus.
503 * Added format attribute negotiation for the Opus codec. Format attribute
504 negotiation is provided by the res_format_attr_opus module.
509 * Masquerades as an operation inside Asterisk have been effectively hidden
510 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
511 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
512 dropping of frame/audio hooks, and other internal implementation details
513 that users had to deal with. This fundamental change has large implications
514 throughout the changes documented for this version. For more information
515 about the new core architecture of Asterisk, please see the Asterisk wiki.
517 * Multiple parties in a bridge may now be transferred. If a participant in a
518 multi-party bridge initiates a blind transfer, a Local channel will be used
519 to execute the dialplan location that the transferer sent the parties to. If
520 a participant in a multi-party bridge initiates an attended transfer,
521 several options are possible. If the attended transfer results in a transfer
522 to an application, a Local channel is used. If the attended transfer results
523 in a transfer to another channel, the resulting channels will be merged into
526 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
527 driver specific. If the channel variable is set on the transferrer channel,
528 the sound will be played to the target of an attended transfer.
530 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
531 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
532 listed. Any more peers in the bridge will not be included in the list.
533 BRIDGEPEER is not valid in holding bridges like parking since those channels
534 do not talk to each other even though they are in a bridge.
536 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
537 and will contain a value if the BRIDGEPEER's channel driver supports it.
539 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
540 was responsible for an attended transfer in a similar fashion to
543 * Modules using the Configuration Framework or Sorcery must have XML
544 configuration documentation. This configuration documentation is included
545 with the rest of Asterisk's XML documentation, and is accessible via CLI
546 commands. See the CLI changes for more information.
548 AMI (Asterisk Manager Interface)
550 * Major changes were made to both the syntax as well as the semantics of the
551 AMI protocol. In particular, AMI events have been substantially improved
552 in this version of Asterisk. For more information, please see the AMI
553 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
555 * AMI events that reference a particular channel or bridge will now always
556 contain a standard set of fields. When multiple channels or bridges are
557 referenced in an event, fields for at least some subset of the channels
558 and bridges in the event will be prefixed with a descriptive name to avoid
559 name collisions. See the AMI event documentation on the Asterisk wiki for
562 * The CLI command 'manager show commands' no longer truncates command names
563 longer than 15 characters and no longer shows authorization requirement
564 for commands. 'manager show command' now displays the privileges needed
565 for using a given manager command instead.
567 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
568 peer in its response if the peer has a subscribe context set.
570 * The SIPqualifypeer action now acknowledges the request once it has
571 established that the request is against a known peer. It also issues a new
572 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
574 * The PlayDTMF action now supports an optional 'Duration' parameter. This
575 specifies the duration of the digit to be played, in milliseconds.
577 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
578 updates when changes occur instead of requiring the use of pollmailboxes.
580 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
581 AMI client to manipulate audio currently being played back on a channel. The
582 supported operations depend on the application being used to send audio to
583 the channel. When the audio playback was initiated using the ControlPlayback
584 application or CONTROL STREAM FILE AGI command, the audio can be paused,
585 stopped, restarted, reversed, or skipped forward. When initiated by other
586 mechanisms (such as the Playback application), the audio can be stopped,
587 reversed, or skipped forward.
589 * Channel related events now contain a snapshot of channel state, adding new
590 fields to many of these events.
592 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
593 in a future release. Please use the common 'Exten' field instead.
595 * The AMI event 'UserEvent' from app_userevent now contains the channel state
596 fields. The channel state fields will come before the body fields.
598 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
599 'UnParkedCall' have changed significantly in the new res_parking module.
601 The 'Channel' and 'From' headers are gone. For the channel that was parked
602 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
603 has a number of fields associated with it. The old 'Channel' header relayed
604 the same data as the new 'ParkeeChannel' header.
606 The 'From' field was ambiguous and changed meaning depending on the event.
607 for most of these, it was the name of the channel that parked the call
608 (the 'Parker'). There is no longer a header that provides this channel name,
609 however the 'ParkerDialString' will contain a dialstring to redial the
610 device that parked the call.
612 On UnParkedCall events, the 'From' header would instead represent the
613 channel responsible for retrieving the parkee. It receives a channel
614 snapshot labeled 'Retriever'. The 'from' field is is replaced with
617 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
619 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
620 fashion has changed the field names 'StartExten' and 'StopExten' to
621 'StartSpace' and 'StopSpace' respectively.
623 * The deprecated use of | (pipe) as a separator in the channelvars setting in
624 manager.conf has been removed.
626 * Channel Variables conveyed with a channel no longer contain the name of the
627 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
628 ChanVariable: bar=baz. When multiple channels are present in a single AMI
629 event, the various ChanVariable fields will contain a suffix that specifies
630 which channel they correspond to.
632 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
633 event always conveys the AMI event for a particular channel.
635 * All 'Reload' events have been consolidated into a single event type. This
636 event will always contain a Module field specifying the name of the module
637 and a Status field denoting the result of the reload. All modules now issue
638 this event when being reloaded.
640 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
641 fail to receive this event due to being connected after modules have loaded.
642 AMI connections that want to know when Asterisk is ready should listen for
643 the 'FullyBooted' event.
645 * app_fax now sends the same send fax/receive fax events as res_fax. The
646 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
647 now the 'ReceiveFAX' event.
649 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
650 'MusicOnHoldStop'. The sub type field has been removed.
652 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
653 carrier for another protocol.
655 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
656 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
657 to the specific channel. 'Both' may be specified to play a tone to both
658 channels. The old 'yes' option is still accepted as a way of playing the
659 tone to Channel2 only.
661 * The AMI 'Status' response event to the AMI Status action replaces the
662 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
663 indicate what bridge the channel is currently in.
665 * The AMI 'Hold' event has been moved out of individual channel drivers, into
666 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
669 * The AMI events in app_queue have been made more consistent with each other.
670 Events that reference channels (QueueCaller* and Agent*) will show
671 information about each channel. The (infamous) 'Join' and 'Leave' AMI
672 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
674 * The 'MCID' AMI event now publishes a channel snapshot when available and
675 its non-channel-snapshot parameters now use either the "MCallerID" or
676 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
677 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
678 parameters in the channel snapshot.
680 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
681 'AgentLogin' and 'AgentLogoff' respectively.
683 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
684 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
686 * 'ChannelUpdate' events have been removed.
688 * All AMI events now contain a 'SystemName' field, if available.
690 * Local channel optimization is now conveyed in two events:
691 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
692 when the Local channel driver begins attempting to optimize itself out of
693 the media path; the End event is sent after the channel halves have
694 successfully optimized themselves out of the media path.
696 * Local channel information in events is now prefixed with 'LocalOne' and
697 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
698 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
699 and 'LocalOptimizationEnd' events.
701 * The option 'allowmultiplelogin' can now be set or overriden in a particular
702 account. When set in the general context, it will act as the default
703 setting for defined accounts.
705 * The 'BridgeAction' event was removed. It technically added no value, as the
706 Bridge Action already receives confirmation of the bridge through a
707 successful completion Event.
709 * The 'BridgeExec' events were removed. These events duplicated the events that
710 occur in the Briding API, and are conveyed now through BridgeCreate,
711 BridgeEnter, and BridgeLeave events.
713 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
714 previous versions. They now report all SR/RR packets sent/received, and
715 have been restructured to better reflect the data sent in a SR/RR. In
716 particular, the event structure now supports multiple report blocks.
718 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
719 raised when a blind transfer/attended transfer completes successfully.
720 They contain information about the transfer that just completed, including
721 the location of the transfered channel.
723 * Added a 'security' class to AMI which outputs the required fields for
724 security messages similar to the log messages from res_security_log
726 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
727 that describes the status value in a human readable string.
729 CDR (Call Detail Records)
731 * Significant changes have been made to the behavior of CDRs. The CDR engine
732 was effectively rewritten and built on the Stasis message bus. For a full
733 definition of CDR behavior in Asterisk 12, please read the specification
734 on the Asterisk wiki (wiki.asterisk.org).
736 * CDRs will now be created between all participants in a bridge. For each
737 pair of channels in a bridge, a CDR is created to represent the path of
738 communication between those two endpoints. This lets an end user choose who
739 to bill for what during bridge operations with multiple parties.
741 * The duration, billsec, start, answer, and end times now reflect the times
742 associated with the current CDR for the channel, as opposed to a cumulative
743 measurement of all CDRs for that channel.
745 * When a CDR is dispatched, user defined CDR variables from both parties are
746 included in the resulting CDR. If both parties have the same variable, only
747 the Party A value is provided.
749 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
750 information regarding the CDR engine is logged as verbose messages. This
751 option should only be used if the behavior of the CDR engine needs to be
754 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
755 normally configured in cdr.conf.
757 * Added CLI command 'cdr show active {channel}'. When {channel} is not
758 specified, this command provides a summary of the channels with CDR
759 information and their statistics. When {channel} is specified, it shows
760 detailed information about all records associated with {channel}.
762 CEL (Channel Event Logging)
764 * CEL has undergone significant rework in Asterisk 12, and is now built on the
765 Stasis message bus. Please see the specification for CEL on the Asterisk
766 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
769 * The 'extra' field of all CEL events that use it now consists of a JSON blob
770 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
772 * BLINDTRANSFER events now report the transferee bridge unique
773 identifier, extension, and context in a JSON blob as the extra string
774 instead of the transferee channel name as the peer.
776 * ATTENDEDTRANSFER events now report the peer as NULL and additional
777 information in the 'extra' string as a JSON blob. For transfers that occur
778 between two bridged channels, the 'extra' JSON blob contains the primary
779 bridge unique identifier, the secondary channel name, and the secondary
780 bridge unique identifier. For transfers that occur between a bridged channel
781 and a channel running an app, the 'extra' JSON blob contains the primary
782 bridge unique identifier, the secondary channel name, and the app name.
784 * LOCAL_OPTIMIZE events have been added to convey local channel
785 optimizations with the record occurring for the semi-one channel and
786 the semi-two channel name in the peer field.
788 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
789 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
790 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
791 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
792 regardless of whether or not that bridge happens to contain multiple
797 * When compiled with '--enable-dev-mode', the astobj2 library will now add
798 several CLI commands that allow for inspection of ao2 containers that
799 register themselves with astobj2. The CLI commands are 'astobj2 container
800 dump', 'astobj2 container stats', and 'astobj2 container check'.
802 * Added specific CLI commands for bridge inspection. This includes 'bridge
803 show all', which lists all bridges in the system, and 'bridge show {id}',
804 which provides specific information about a bridge.
806 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
807 ejecting the channels currently in the bridge. If the channels cannot
808 continue in the dialplan or application that put them in the bridge, they
811 * Added command 'bridge kick'. This will eject a single channel from a bridge.
813 * Added commands to inspect and manipulate the registered bridge technologies.
814 This include 'bridge technology show', which lists the registered bridge
815 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
816 which controls whether or not a registered bridge technology can be used
817 during smart bridge operations. If a technology is suspended, it will not
818 be used when a bridge technology is picked for channels; when unsuspended,
819 it can be used again.
821 * The command 'config show help {module} {type} {option}' will show
822 configuration documentation for modules with XML configuration
823 documentation. When {module}, {type}, and {option} are omitted, a listing
824 of all modules with registered documentation is displayed. When {module}
825 is specified, a listing of all configuration types for that module is
826 displayed, along with their synopsis. When {module} and {type} are
827 specified, a listing of all configuration options for that type are
828 displayed along with their synopsis. When {module}, {type}, and {option}
829 are specified, detailed information for that configuration option is
832 * Added 'core show sounds' and 'core show sound' CLI commands. These display
833 a listing of all installed media sounds available on the system and
834 detailed information about a sound, respectively.
836 * 'xmldoc dump' has been added. This CLI command will dump the XML
837 documentation DOM as a string to the specified file. The Asterisk core
838 will populate certain XML elements pulled from the source files with
839 additional run-time information; this command lets a user produce the
840 XML documentation with all information.
844 * Parking has been pulled from core and placed into a separate module called
845 res_parking. See Parking changes below for more details. Configuration for
846 parking should now be performed in res_parking.conf. Configuration for
847 parking in features.conf is now unsupported.
849 * Core attended transfers now have several new options. While performing an
850 attended transfer, the transferer now has the following options:
851 - *1 - cancel the attended transfer (configurable via atxferabort)
852 - *2 - complete the attended transfer, dropping out of the call
853 (configurable via atxfercomplete)
854 - *3 - complete the attended transfer, but stay in the call. This will turn
855 the call into a multi-party bridge (configurable via atxferthreeway)
856 - *4 - swap to the other party. Once an attended transfer has begun, this
857 options may be used multiple times (configurable via atxferswap)
859 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
860 must be on the channel initiating the transfer to have any effect.
862 * The BRIDGE_FEATURES channel variable would previously only set features for
863 the calling party and would set this feature regardless of whether the
864 feature was in caps or in lowercase. Use of a caps feature for a letter
865 will now apply the feature to the calling party while use of a lowercase
866 letter will apply that feature to the called party.
868 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
870 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
871 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
872 activated the dynamic feature.
874 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
875 only on the channel executing the dynamic feature. Executing a dynamic
876 feature on the bridge peer in a multi-party bridge will execute it on all
877 peers of the activating channel.
879 * You can now have the settings for a channel updated using the FEATURE()
880 and FEATUREMAP() functions inherited to child channels by setting
881 FEATURE(inherit)=yes.
883 * automixmon now supports additional channel variables from automon including:
884 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
885 and TOUCH_MIXMONITOR_MESSAGE_STOP
887 * A new general features.conf option 'recordingfailsound' has been added which
888 allowssetting a failure sound for a user tries to invoke a recording feature
889 such as automon or automixmon and it fails.
891 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
892 features.c for atxferdropcall=no to work properly. This option now just
897 * Added log rotation strategy 'none'. If set, no log rotation strategy will
898 be used. Given that this can cause the Asterisk log files to grow quickly,
899 this option should only be used if an external mechanism for log management
904 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
905 will store the path information for that peer when it registers. Realtime
906 tables can also use the 'supportpath' field to enable Path header support.
908 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
909 objectIdentifier. This maps to the supportpath option in sip.conf.
913 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
914 provides modules a useful abstraction on top of the many storage mechanisms
915 in Asterisk, including the Asterisk Database, static configuration files,
916 static Realtime, and dynamic Realtime. It also provides a caching service.
917 Users can configure a hierarchy of data storage layers for specific modules
920 * All future modules which utilize Sorcery for object persistence must have a
921 column named "id" within their schema when using the Sorcery realtime module.
922 This column must be able to contain a string of up to 128 characters in length.
924 Security Events Framework
926 * Security Event timestamps now use ISO 8601 formatted date/time instead of
927 the "seconds-microseconds" format that it was using previously.
931 * The Stasis message bus is a publish/subscribe message bus internal to
932 Asterisk. Many services in Asterisk are built on the Stasis message bus,
933 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
934 Stasis can be configured in stasis.conf. Note that these parameters operate
935 at a very low level in Asterisk, and generally will not require changes.
939 * When a channel driver is configured to enable jiterbuffers, they are now
940 applied unconditionally when a channel joins a bridge. If a jitterbuffer
941 is already set for that channel when it enters, such as by the JITTERBUFFER
942 function, then the existing jitterbuffer will be used and the one set by
943 the channel driver will not be applied.
947 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
948 dialplan applications provided by the app_agent_pool module. Agents are
949 connected with callers using the new AgentRequest dialplan application.
950 The Agents:<agent-id> device state is available to monitor the status of an
951 agent. See agents.conf.sample for valid configuration options.
953 * The updatecdr option has been removed. Altering the names of channels on a
954 CDR is not supported - the name of the channel is the name of the channel,
955 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
956 has also been removed, for the same reason.
958 * The endcall and enddtmf configuration options are removed. Use the
959 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
960 channel before calling AgentLogin.
964 * chan_bridge has been removed. Its functionality has been incorporated
965 directly into the ConfBridge application itself.
969 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
970 of the specified span and its B-channels. Note that this command should
971 only be used if you understand the risks it entails.
973 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
974 A range of channels can be specified to be destroyed. Note that this command
975 should only be used if you understand the risks it entails.
977 * Added the CLI command 'dahdi create channels'. A range of channels can be
978 specified to be created, or the keyword 'new' can be used to add channels
981 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
982 the exact configured mailbox name. For app_voicemail mailboxes this is
985 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
989 * IPv6 support has been added. We are now able to bind to and
990 communicate using IPv6 addresses.
994 * The /b option has been removed.
996 * chan_local moved into the system core and is no longer a loadable module.
1000 * Added general support for busy detection.
1002 * Added ECAM command support for Sony Ericsson phones.
1006 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1007 SIP stack. A collection of resource modules provides the bulk of the SIP
1008 functionality. For more information on the new SIP channel driver, see
1009 https://wiki.asterisk.org/wiki/x/JYGLAQ
1013 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1014 using the 'supportpath' setting, either on a global basis or on a peer basis.
1015 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1016 set of proxies by using a pre-loaded route-set defined by the Path headers in
1017 the REGISTER request. See Realtime updates for more configuration information.
1019 * The SIP_CODEC family of variables may now specify more than one codec. Each
1020 codec must be separated by a comma. The first codec specified is the
1021 preferred codec for the offer. This allows a dialplan writer to specify both
1022 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1024 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1025 in the core, and can be filtered out using the 'eventfilter' parameter
1028 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1029 codecs configured for a peer instead of the requested codec.
1031 * The option "register_retry_403" has been added to chan_sip to work around
1032 servers that are known to erroneously send 403 in response to valid
1033 REGISTER requests and allows Asterisk to continue attepmting to connect.
1037 * Added the 'immeddialkey' parameter. If set, when the user presses the
1038 configured key the already entered number will be immediately dialed. This
1039 is useful when the dialplan allows for variable length pattern matching.
1040 Valid options are '*' and '#'.
1042 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1043 milliseconds) before a call forward is considered to not be answered.
1045 * The 'serviceurl' parameter allows Service URLs to be attached to line
1054 * The password option has been disabled, as the AgentLogin application no
1055 longer provides authentication.
1059 * Due to changes in the Asterisk core, this function is no longer needed to
1060 preserve a MixMonitor on a channel during transfer operations and dialplan
1061 execution. It is effectively obsolete.
1065 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1066 deprecated. Use the CHANNEL function instead to access these attributes.
1068 * The 'l' option has been removed. When reading a CDR attribute, the most
1069 recent record is always used. When writing a CDR attribute, all non-finalized
1072 * The 'r' option has been removed, for the same reason as the 'l' option.
1074 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1079 * A new function CDR_PROP has been added. This function lets you set properties
1080 on a channel's active CDRs. This function is write-only. Properties accept
1081 boolean values to set/clear them on the channel's CDRs. Valid properties
1083 - 'party_a' - make this channel the preferred Party A in any CDR between two
1084 channels. If two channels have this property set, the creation time of the
1085 channel is used to determine who is Party A. Note that dialed channels are
1086 never Party A in a CDR.
1087 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1088 application when set to True, and analogous to the 'e' option in ResetCDR
1093 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1094 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1095 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1098 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1099 string, i.e., [[context],extension],priority. If set on a channel, if a
1100 channel leaves a bridge but is not hung up it will resume dialplan execution
1105 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1106 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1107 The value of this setting is ignored when disabled is used for the argument.
1111 * A new function provided by chan_pjsip, this function can be used in
1112 conjunction with the Dial application to construct a dial string that will
1113 dial all contacts on an Address of Record associated with a chan_pjsip
1118 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1119 outbound channel prior to dialing.
1123 * Redirecting reasons can now be set to arbitrary strings. This means
1124 that the REDIRECTING dialplan function can be used to set the redirecting
1125 reason to any string. It also allows for custom strings to be read as the
1126 redirecting reason from SIP Diversion headers.
1130 * The SPEECH_ENGINE function now supports read operations. When read from, it
1131 will return the current value of the requested attribute.
1135 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1136 system as mailbox@context. The rest of the system cannot add @default
1137 to mailbox identifiers for app_voicemail that do not specify a context
1138 any longer. It is a mailbox identifier format that should only be
1139 interpreted by app_voicemail.
1145 res_agi (Asterisk Gateway Interface)
1147 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1149 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1152 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1153 will start the playback of the audio at the position specified. It will
1154 also return the final position of the file in 'endpos'.
1156 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1157 channel variable if the user stopped the file playback or if a remote
1158 entity stopped the playback. If neither stopped the playback, it will
1159 indicate the overall success/failure of the playback. If stopped early,
1160 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1163 * The SAY ALPHA command now accepts an additional parameter to control
1164 whether it specifies the case of uppercase, lowercase, or all letters to
1165 provide functionality similar to SayAlphaCase.
1167 res_ari (Asterisk RESTful Interface) (and others)
1169 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1170 control telephony primitives in Asterisk by remote client. This includes
1171 channels, bridges, endpoints, media, and other fundamental concepts. Users
1172 of ARI can develop their own communications applications, controlling
1173 multiple channels using an HTTP RESTful interface and receiving JSON events
1174 about the objects via a WebSocket connection. ARI can be configured in
1175 Asterisk via ari.conf. For more information on ARI, see
1176 https://wiki.asterisk.org/wiki/x/0YCLAQ
1180 * Parking has been extracted from the Asterisk core as a loadable module,
1181 res_parking. Configuration for parking is now provided by res_parking.conf.
1182 Configuration through features.conf is no longer supported.
1184 * res_parking uses the configuration framework. If an invalid configuration is
1185 supplied, res_parking will fail to load or fail to reload. Previously,
1186 invalid configurations would generally be accepted, with certain errors
1187 resulting in individually disabled parking lots.
1189 * Parked calls are now placed in bridges. While this is largely an
1190 architectural change, it does have implications on how channels in a parking
1191 lot are viewed. For example, commands that display channels in bridges will
1192 now also display the channels in a parking lot.
1194 * The order of arguments for the new parking applications have been modified.
1195 Timeout and return context/exten/priority are now implemented as options,
1196 while the name of the parking lot is now the first parameter. See the
1197 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1198 in-depth information as well as syntax.
1200 * Extensions are by default no longer automatically created in the dialplan to
1201 park calls or pickup parked calls. Generation of dialplan extensions can be
1202 enabled using the 'parkext' configuration option.
1204 * ADSI functionality for parking is no longer supported. The 'adsipark'
1205 configuration option has been removed as a result.
1207 * The PARKINGSLOT channel variable has been deprecated in favor of
1208 PARKING_SPACE to match the naming scheme of the new system.
1210 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1211 channel even when the configuration option 'comebactoorigin' is enabled.
1213 * A new CLI command 'parking show' has been added. This allows a user to
1214 inspect the parking lots that are currently in use.
1215 'parking show <parkinglot>' will also show the parked calls in a specific
1218 * The CLI command 'parkedcalls' is now deprecated in favor of
1219 'parking show <parkinglot>'.
1221 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1222 can be used to get a list of parked calls for a specific parking lot.
1224 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1225 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1226 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1227 longer a required argument.
1229 * The ParkAndAnnounce application is now provided through res_parking instead
1230 of through the separate app_parkandannounce module.
1232 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1233 by default. Instead, it will follow the timeout rules of the parking lot. The
1234 old behavior can be reproduced by using the 'c' option.
1236 * Dynamic parking lots will now fail to be created under the following
1238 - if the parking lot specified by PARKINGDYNAMIC does not exist
1239 - if they require exclusive park and parkedcall extensions which overlap
1240 with existing parking lots.
1242 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1243 currently contain no calls. Dynamic parking lots containing parked calls
1244 will persist through the reloads without alteration.
1246 * If 'parkext_exclusive' is set for a parking lot and that extension is
1247 already in use when that parking lot tries to register it, this is now
1248 considered a parking system configuration error. Configurations which do
1249 this will be rejected.
1251 * Added channel variable PARKER_FLAT. This contains the name of the extension
1252 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1253 comebacktoorigin is disabled, but the dialplan or an external control
1254 mechanism wants to use the extension in the park-dial context that was
1255 generated to re-dial the parker on timeout.
1257 res_pjsip (and many others)
1259 * A large number of resource modules make up the SIP stack based on pjsip.
1260 The chan_pjsip channel driver users these resource modules to provide
1261 various SIP functionality in Asterisk. The majority of configuration for
1262 these modules is performed in pjsip.conf. Other modules may use their
1263 own configuration files.
1265 * Added 'set_var' option for an endpoint. For each variable specified that
1266 variable gets set upon creation of a channel involving the endpoint.
1270 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1271 them, an Asterisk-specific version of PJSIP needs to be installed.
1272 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1274 res_statsd/res_chan_stats
1276 * A new resource module, res_statsd, has been added, which acts as a statsd
1277 client. This module allows Asterisk to publish statistics to a statsd
1278 server. In conjunction with res_chan_stats, it will publish statistics about
1279 channels to the statsd server. It can be configured via res_statsd.conf.
1283 * Device state for XMPP buddies is now available using the following format:
1284 XMPP/<client name>/<buddy address>
1285 If any resource is available the device state is considered to be not in use.
1286 If no resources exist or all are unavailable the device state is considered
1293 Realtime/Database Scripts
1295 * Asterisk previously included example db schemas in the contrib/realtime/
1296 directory of the source tree. This has been replaced by a set of database
1297 migrations using the Alembic framework. This allows you to use alembic to
1298 initialize the database for you. It will also serve as a database migration
1299 tool when upgrading Asterisk in the future.
1301 See contrib/ast-db-manage/README.md for more details.
1305 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1306 This python script will convert an existing sip.conf file to a
1307 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1308 is meant to be an aid in converting an existing chan_sip configuration to
1309 a chan_pjsip configuration, but it is expected that configuration beyond
1310 what the script provides will be needed.
1313 ------------------------------------------------------------------------------
1314 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1315 ------------------------------------------------------------------------------
1319 * The Asterisk build system will now build and install a shared library
1320 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1321 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1322 that Asterisk can ensure that these functions do *not* get called by any
1323 modules that are loaded into Asterisk, since they should only be called once
1324 in any single process. If desired, this feature can be disabled by supplying
1325 the "--disable-asteriskssl" option to the configure script.
1327 * A new make target, 'full', has been added to the Makefile. This performs
1328 the same compilation actions as make all, but will also scan the entirety of
1329 each source file for documentation. This option is needed to generate AMI
1330 event documentation. Note that your system must have Python in order for
1331 this make target to succeed.
1333 * The optimization portion of the build system has been reworked to avoid
1334 broken builds on certain architectures. All architecture-specific
1335 optimization has been removed in favor of using -march=native to allow gcc
1336 to detect the environment in which it is running when possible. This can
1337 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1339 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1340 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1342 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1343 previously parsed the header file to obtain the version of Asterisk, you
1344 will now have to go through Asterisk to get the version information.
1352 * Added 'F()' option. Similar to the dial option, this can be supplied with
1353 arguments indicating where the callee should go after the caller is hung up,
1354 or without options specified, the priority after the Queue will be used.
1359 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1360 all non-admin participants on a conference. The confbridge configuration
1361 file also allows for the default sounds played to all conference users when
1362 this occurs to be overriden using sound_participants_unmuted and
1363 sound_participants_muted.
1365 * Added menu action participant_count. This will playback the number of
1366 current participants in a conference.
1368 * Added announcement configuration option to user profile. If set the sound
1369 file will be played to the user, and only the user, upon joining the
1372 * Added record_file_append option that defaults to "yes", but if set to no
1373 will create a new file between each start/stop recording.
1378 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1379 channels respectively before the callee channels are called.
1384 * Added support for IPv6.
1386 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1387 external process will cause the current playlist to be cleared, including
1388 stopping any audio file that is currently playing. This is useful when you
1389 want to interrupt audio playback only when specific DTMF is entered by the
1395 * A new option, 'I' has been added to app_followme. By setting this option,
1396 Asterisk will not update the caller with connected line changes when they
1397 occur. This is similar to app_dial and app_queue.
1399 * The 'N' option is now ignored if the call is already answered.
1401 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1402 and caller channels respectively before the callee channels are called.
1404 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1410 * MixMonitor hooks now have IDs associated with them which can be used to
1411 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1412 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1413 now accepts that ID as an argument.
1415 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1416 indicated mailboxes.
1421 * The connect action in app_mysql now allows you to specify a port number to
1422 connect to. This is useful if you run a MySQL server on a non-standard
1428 * Increased the default number of allowed destinations from 5 to 12.
1433 * The app_page application now no longer depends on DAHDI or app_meetme. It
1434 has been re-architected to use app_confbridge internally.
1439 * Added queue options autopausebusy and autopauseunavail for automatically
1440 pausing a queue member when their device reports busy or congestion.
1442 * The 'ignorebusy' option for queue members has been deprecated in favor of
1443 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1444 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1445 per interface basis. Individual ringinuse values can now be set in
1446 queues.conf via an argument to member definitions. Lastly, the queue
1447 'ringinuse' setting now only determines defaults for the per member
1448 'ringinuse' setting and does not override per member settings like it does
1449 in earlier versions.
1451 * Added 'F()' option. Similar to the dial option, this can be supplied with
1452 arguments indicating where the callee should go after the caller is hung up,
1453 or without options specified, the priority after the Queue will be used.
1455 * Added new option log_member_name_as_agent, which will cause the membername to
1456 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1457 state_interface has been set.
1459 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1461 * App_queue will now play periodic announcements for the caller that
1462 holds the first position in the queue while waiting for answer.
1466 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1467 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1468 changed arguments to SayUnixTime so that every option is truly optional even
1469 when using multiple options (so that j option could be used without having to
1470 manually specify timezone and format) There are other benefits, e.g., format
1471 can now be used without specifying time zone as well.
1476 * Addition of the VM_INFO function - see Function changes.
1478 * The imapserver, imapport, and imapflags configuration options can now be
1479 overriden on a user by user basis.
1481 * When voicemail plays a message's envelope with saycid set to yes, when
1482 reaching the caller id field it will play a recording of a file with the same
1483 base name as the sender's callerid if there is a similarly named file in
1484 <astspooldir>/recordings/callerids/
1486 * Voicemails now contains a unique message identifier "msg_id", which is stored
1487 in the message envelope with the sound files. IMAP backends will now store
1488 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1489 backends will store the message identifier in a "msg_id" column. See
1490 UPGRADE.txt for more information.
1492 * Added VoiceMailPlayMsg application. This application will play a single
1493 voicemail message from a mailbox. The result of the application, SUCCESS or
1494 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1499 * Hangup handlers can be attached to channels using the CHANNEL() function.
1500 Hangup handlers will run when the channel is hung up similar to the h
1501 extension. The hangup_handler_push option will push a GoSub compatible
1502 location in the dialplan onto the channel's hangup handler stack. The
1503 hangup_handler_pop option will remove the last added location, and optionally
1504 replace it with a new GoSub compatible location. The hangup_handler_wipe
1505 option will remove all locations on the stack, and optionally add a new
1508 * The expression parser now recognizes the ABS() absolute value function,
1509 which will convert negative floating point values to positive values.
1511 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1512 control of faxdetect.
1514 * Addition of the VM_INFO function that can be used to retrieve voicemail
1515 user information, such as the email address and full name.
1516 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1519 * The REDIRECTING function now supports the redirecting original party id
1522 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1523 lets you set some of the configuration options from the [general] section
1524 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1525 the key sequence used to activate built-in features, such as blindxfer,
1526 and automon. See the built-in documentation for details.
1528 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1529 instead of simply the uri. This is the format that MessageSend() can use
1530 in the from parameter for outgoing SIP messages.
1532 * Added the PRESENCE_STATE function. This allows retrieving presence state
1533 information from any presence state provider. It also allows setting
1534 presence state information from a CustomPresence presence state provider.
1535 See AMI/CLI changes for related commands.
1537 * Added the AMI_CLIENT function to make manager account attributes available
1538 to the dialplan. It currently supports returning the current number of
1539 active sessions for a given account.
1541 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1542 and the REDIRECTING functions.
1550 * Added a manager event "LocalBridge" for local channel call bridges between
1551 the two pseudo-channels created.
1556 * Added dialtone_detect option for analog ports to disconnect incoming
1557 calls when dialtone is detected.
1559 * Added option colp_send to send ISDN connected line information. Allowed
1560 settings are block, to not send any connected line information; connect, to
1561 send connected line information on initial connect; and update, to send
1562 information on any update during a call. Default is update.
1564 * Add options namedcallgroup and namedpickupgroup to support installations
1565 where a higher number of groups (>64) is required.
1567 * Added support to use private party ID information with PRI calls.
1572 * A new channel driver named chan_motif has been added which provides support for
1573 Google Talk and Jingle in a single channel driver. This new channel driver includes
1574 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1575 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1576 specification, current Google Jingle specification, and the original Google Talk
1582 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1583 globally and overriden on a peer by peer basis.
1585 * Direct media functionality has been added. Options in config are:
1586 directmedia (directrtp) and directrtpsetup (earlydirect)
1588 * ChannelUpdate events now contain a CallRef header.
1593 * Asterisk will no longer substitute CID number for CID name in the display
1594 name field if CID number exists without a CID name. This change improves
1595 compatibility with certain device features such as Avaya IP500's directory
1598 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1599 created using that setting to not be removed during SIP reload.
1601 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1602 request with a "Record:" header, this will turn the requested feature on/off.
1603 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1604 dynamic features must be enabled and configured properly on the requesting
1605 channel for this to function properly.
1607 * Add support to realtime for the 'callbackextension' option.
1609 * When multiple peers exist with the same address, but differing
1610 callbackextension options, incoming requests that are matched by address
1611 will be matched to the peer with the matching callbackextension if it is
1614 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1615 which set the force_rport and comedia options automatically if Asterisk
1616 detects that an incoming SIP request crossed a NAT after being sent by
1617 the remote endpoint.
1619 * The default global nat setting in sip.conf has been changed from force_rport
1620 to auto_force_rport.
1622 * NAT settings are now a combinable list of options. The equivalent of the
1623 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1625 * Adds an option send_diversion which can be disabled to prevent
1626 diversion headers from automatically being added to INVITE requests.
1628 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1629 be sent to the remote host at a given interval to keep the NAT mapping open.
1630 This can be enabled using the keepalive configuration option.
1632 * Add option 'tonezone' to specify country code for indications. This option
1633 can be set both globally and overridden for specific peers.
1635 * The SIP Security Events Framework now supports IPv6.
1637 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1638 between multiple user agents. When set, for directmedia reinvites,
1639 Asterisk will not send an immediate reinvite on an incoming call leg. This
1640 option is useful when peered with another SIP user agent that is known to
1641 send immediate direct media reinvites upon call establishment.
1643 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1646 * Add options subminexpiry and submaxexpiry to set limits of subscription
1647 timer independently from registration timer settings. The setting of the
1648 registration timer limits still is done by options minexpiry, maxexpiry
1649 and defaultexpiry. For backwards compatibility the setting of minexpiry
1650 and maxexpiry also is used to configure the subscription timer limits if
1651 subminexpiry and submaxexpiry are not set in sip.conf.
1653 * Set registration timer limits to default values when reloading sip
1654 configuration and values are not set by configuration.
1656 * Add options namedcallgroup and namedpickupgroup to support installations
1657 where a higher number of groups (>64) is required.
1659 * When a MESSAGE request is received, the address the request was received from
1660 is now saved in the SIP_RECVADDR variable.
1662 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1663 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1664 the ANI2/OLI information is set on the channel, which can be retrieved using
1665 the CALLERID function.
1667 * Peers can now be configured to support negotiation of ICE candidates using
1668 the setting icesupport. See res_rtp_asterisk changes for more information.
1670 * Added support for format attribute negotiation. See the Codecs changes for
1673 * Extra headers specified with SIPAddHeader are sent with the REFER message
1674 when using Transfer application. See refer_addheaders in sip.conf.sample.
1676 * Added support to use private party ID information with calls.
1678 * Adds an option discard_remote_hold_retrieval that when set stops telling
1679 the peer to start music on hold.
1684 * Added skinny version 17 protocol support.
1688 --------------------
1689 * Added ability to use multiple lines for a single phone. This allows multiple
1690 calls to occur on a single phone, using callwaiting and switching between calls.
1692 * Added option 'sharpdial' allowing end dialing by pressing # key
1694 * Added option 'interdigit_timer' to control phone dial timeout
1696 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1698 * Added global 'debug' option, that enables debug in channel driver
1700 * Added ability to translate on-screen menu in multiple languages. Tested on
1701 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1702 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1705 * In addition to English added French and Russian languages for on-screen menus
1707 * Reworked dialing number input: added dialing by timeout, immediate dial on
1708 on dialplan compare, phone number length now not limited by screen size
1710 * Added ability to pickup a call using features.conf defined value and
1716 * Add options namedcallgroup and namedpickupgroup to support installations
1717 where a higher number of groups (>64) is required.
1719 * Added support to use private party ID information with calls.
1724 * The minimum DTMF duration can now be configured in asterisk.conf
1725 as "mindtmfduration". The default value is (as before) set to 80 ms.
1726 (previously it was only available in source code)
1728 * Named ACLs can now be specified in acl.conf and used in configurations that
1729 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1730 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1731 working ACL. In addition, some CLI commands have been added to provide
1732 show information and allow for module reloading - see CLI Changes.
1734 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1735 items (separated by commas), and items in the rule can be negated by prefixing
1736 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1737 longer necessray to control the order that the 'permit' and 'deny' columns are
1738 returned from queries.
1740 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1741 be used within the dynamic weight attribute when specifying a mapping.
1743 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1744 header, instead of putting the user defined event name there. When enabled
1745 the UserDefType header is added for user defined events. This feature is
1746 enabled with the setting show_user_defined.
1748 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1749 line purposes use the following variables instead of their macro equivalents:
1750 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1751 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1752 cc_callback_macro in channel configurations.
1754 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1757 * Call files now support the "early_media" option to connect with an outgoing
1758 extension when early media is received.
1760 * Added support to use private party ID information with calls.
1765 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1766 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1767 AGI application would exit immediately after a channel hangup is detected.
1769 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1770 are resolved and each address is attempted in turn until one succeeds or
1774 AMI (Asterisk Manager Interface)
1776 * The originate action now has an option "EarlyMedia" that enables the
1777 call to bridge when we get early media in the call. Previously,
1778 early media was disregarded always when originating calls using AMI.
1780 * Added setvar= option to manager accounts (much like sip.conf)
1782 * Originate now generates an error response if the extension given is not found
1785 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1786 them if the i(variable) option is used. StopMixMonitor will accept
1787 MixMonitorID as an option to close specific MixMonitors.
1789 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1790 updated to include information about peers configured with
1791 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1792 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1793 returned if auto_force_rport is not enabled.
1795 * Added SIPpeerstatus manager command which will generate PeerStatus events
1796 similar to the existing PeerStatus events found in chan_sip on demand.
1798 * Hangup now can take a regular expression as the Channel option. If you want
1799 to hangup multiple channels, use /regex/ as the Channel option. Existing
1800 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1801 the manager will send you a list of channels back that were hung up.
1803 * Support for IPv6 addresses has been added.
1805 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1806 documentation is only generated when Asterisk is compiled using 'make full'.
1807 See the CLI section for commands to display AMI event information.
1809 * The AMI Hangup event now includes the AccountCode header so you can easily
1810 correlate with AMI Newchannel events.
1812 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1813 the StateInterface of the queue member.
1815 * Added AMI event SessionTimeout in the Call category that is issued when a
1816 call is terminated due to either RTP stream inactivity or SIP session timer
1819 * CEL events can now contain a user defined header UserDefType. See core
1820 changes for more information.
1822 * OOH323 ChannelUpdate events now contain a CallRef header.
1824 * Added PresenceState command. This command will report the presence state for
1825 the given presence provider.
1827 * Added Parkinglots command. This will list all parking lots as a series of
1828 AMI Parkinglot events.
1830 * Added MessageSend command. This behaves in the same manner as the
1831 MessageSend application, and is a technolgoy agnostic mechanism to send out
1832 of call text messages.
1834 * Added "message" class authorization. This grants an account permission to
1835 send out of call messages. Write-only.
1840 * The "dialplan add include" command has been modified to create context a context
1841 if one does not already exist. For instance, "dialplan add include foo into bar"
1842 will create context "bar" if it does not already exist.
1844 * A "dialplan remove context" command has been added to remove a context from
1847 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1848 filenames of all running mixmonitors on a channel.
1850 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1851 numeric instead of 0, 1, or 2.
1853 * "stun show status" will show a table describing how the STUN client is
1856 * "acl show [named acl]" will show information regarding a Named ACL. The
1857 acl module can be reloaded with "reload acl".
1859 * Added CLI command to display AMI event information - "manager show events",
1860 which shows a list of all known and documented AMI events, and "manager show
1861 event [event name]", which shows detail information about a specific AMI
1864 * The result of the CLI command "queue show" now includes the state interface
1865 information of the queue member.
1867 * The command "core set verbose" will now set a separate level of logging for
1868 each remote console without affecting any other console.
1870 * Added command "cdr show pgsql status" to check connection status
1872 * "sip show channel" will now display the complete route set.
1874 * Added "presencestate list" command. This command will list all custom
1875 presence states that have been set by using the PRESENCE_STATE dialplan
1878 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1879 command. This changes a custom presence to a new state.
1884 * Codec lists may now be modified by the '!' character, to allow succinct
1885 specification of a list of codecs allowed and disallowed, without the
1886 requirement to use two different keywords. For example, to specify all
1887 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1889 * Add support for parsing SDP attributes, generating SDP attributes, and
1890 passing it through. This support includes codecs such as H.263, H.264, SILK,
1891 and CELT. You are able to set up a call and have attribute information pass.
1892 This should help considerably with video calls.
1894 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1895 just like the GSM codec.
1899 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1900 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1904 * Asterisk version and build information is now logged at the beginning of a
1907 * Threads belonging to a particular call are now linked with callids which get
1908 added to any log messages produced by those threads. Log messages can now be
1909 easily identified as involved with a certain call by looking at their call id.
1910 Call ids may also be attached to log messages for just about any case where
1911 it can be determined to be related to a particular call.
1913 * Each logging destination and console now have an independent notion of the
1914 current verbosity level. Logger.conf now allows an optional argument to
1915 the 'verbose' specifier, indicating the level of verbosity sent to that
1916 particular logging destination. Additionally, remote consoles now each
1917 have their own verbosity level. The command 'core set verbose' will now set
1918 a separate level for each remote console without affecting any other
1924 * Added 'announcement' option which will play at the start of MOH and between
1925 songs in modes of MOH that can detect transitions between songs (eg.
1931 * New per parking lot options: comebackcontext and comebackdialtime. See
1932 configs/features.conf.sample for more details.
1934 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1937 * Channel variable PARKEDCALL is now set with the name of the parking lot
1938 when a timeout occurs.
1944 CDR Postgresql Driver
1946 * Added command "cdr show pgsql status" to check connection status
1949 CDR Adaptive ODBC Driver
1951 * Added schema option for databases that support specifying a schema.
1959 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1960 CALENDAR_WRITE has completed successfully.
1965 * A new option, 'probation' has been added to rtp.conf
1966 RTP in strictrtp mode can now require more than 1 packet to exit learning
1967 mode with a new source (and by default requires 4). The probation option
1968 allows the user to change the required number of packets in sequence to any
1969 desired value. Use a value of 1 to essentially restore the old behavior.
1970 Also, with strictrtp on, Asterisk will now drop all packets until learning
1971 mode has successfully exited. These changes are based on how pjmedia handles
1972 media sources and source changes.
1974 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1975 enabled or disabled using the icesupport setting. A variety of other
1976 settings have been introduced to configure STUN/TURN connections.
1981 * A new module, res_corosync, has been introduced. This module uses the
1982 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1983 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1984 Device State (presence) information. This module is very similar to, and
1985 is a replacement for the res_ais module that was in previous releases of
1991 * This module adds a cleaned up, drop-in replacement for res_jabber called
1992 res_xmpp. This provides the same externally facing functionality but is
1993 implemented differently internally. res_jabber has been deprecated in favor
1994 of res_xmpp; please see the UPGRADE.txt file for more information.
1999 * The safe_asterisk script has been updated to allow several of its parameters
2000 to be set from environment variables. This also enables a custom run
2001 directory of Asterisk to be specified, instead of defaulting to /tmp.
2003 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2004 its value to determine the directory to assume is the top-level directory of
2005 the source tree. If the variable is not set, it defaults to the current
2006 behavior and uses the current working directory.
2008 ------------------------------------------------------------------------------
2009 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2010 ------------------------------------------------------------------------------
2014 * Asterisk now has protocol independent support for processing text messages
2015 outside of a call. Messages are routed through the Asterisk dialplan.
2016 SIP MESSAGE and XMPP are currently supported. There are options in
2017 jabber.conf and sip.conf to allow enabling these features.
2018 -> jabber.conf: see the "sendtodialplan" and "context" options.
2019 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2020 and "outofcall_message_context" options.
2021 The MESSAGE() dialplan function and MessageSend() application have been
2022 added to go along with this functionality. More detailed usage information
2023 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2024 * If real-time text support (T.140) is negotiated, it will be preferred for
2025 sending text via the SendText application. For example, via SIP, messages
2026 that were once sent via the SIP MESSAGE request would be sent via RTP if
2027 T.140 text is negotiated for a call.
2031 * parkedmusicclass can now be set for non-default parking lots.
2033 Asterisk Manager Interface
2034 --------------------------
2035 * PeerStatus now includes Address and Port.
2036 * Added Hold events for when the remote party puts the call on and off hold
2037 for chan_dahdi ISDN channels.
2038 * Added new action MeetmeListRooms to list active conferences (shows same
2039 data as "meetme list" at the CLI).
2040 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2041 Description field that is set by 'description' in the channel configuration
2043 * Added Uniqueid header to UserEvent.
2044 * Added new action FilterAdd to control event filters for the current session.
2045 This requires the system permission and uses the same filter syntax as
2046 filters that can be defined in manager.conf
2047 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2048 versions had some instances of the event converted, but others were left
2049 as-is. All Unlink events should now be converted to Bridge events. The AMI
2050 protocol version number was incremented to 1.2 as a result of this change.
2052 Asterisk HTTP Server
2053 --------------------------
2054 * The HTTP Server can bind to IPv6 addresses.
2057 --------------------------
2058 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2059 with busydetect. usage example: busypattern=200,200,200,600
2062 --------------------------
2063 * New 'gtalk show settings' command showing the current settings loaded from
2065 * The 'logger reload' command now supports an optional argument, specifying an
2066 alternate configuration file to use.
2067 * 'dialplan add extension' command will now automatically create a context if
2068 the specified context does not exist with a message indicated it did so.
2069 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2070 Description field which can be populated with 'description' in the channel
2071 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2074 --------------------------
2075 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2076 thus allowing records which do NOT match the specified filter.
2077 * Added ability to log CONGESTION calls to CDR
2080 --------------------------
2081 * Ability to define custom SILK formats in codecs.conf.
2082 * Addition of speex32 audio format with translation.
2083 * CELT codec pass-through support and ability to define
2084 custom CELT formats in codecs.conf.
2085 * Ability to read raw signed linear files with sample rates
2086 ranging from 8khz - 192khz. The new file extensions introduced
2087 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2088 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2089 Skinny, H.323, etc) can still only support the following codecs:
2090 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2091 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2092 Video: h261, h263, h263p, h264, mpeg4
2097 --------------------------
2098 * New highly optimized and customizable ConfBridge application capable of
2099 mixing audio at sample rates ranging from 8khz-96khz.
2100 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2101 and bridge profiles on a channel.
2102 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2103 about a conference such as locked status and number of parties, admins,
2105 * Addition of video_mode option in confbridge.conf for adding video support
2106 into a bridge profile.
2107 * Addition of the follow_talker video_mode in confbridge.conf. This video
2108 mode dynamically switches the video feed to always display the loudest talker
2109 supplying video in the conference.
2113 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2114 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2115 variables from asterisk.conf.
2119 * Addition of the JITTERBUFFER dialplan function. This function allows
2120 for jitterbuffering to occur on the read side of a channel. By using
2121 this function conference applications such as ConfBridge and MeetMe can
2122 have the rx streams jitterbuffered before conference mixing occurs.
2123 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2125 * Added STRREPLACE function. This function let's the user search a variable
2126 for a given string to replace with another string as many times as the
2127 user specifies or just throughout the whole string.
2128 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2129 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2130 * Added extensions to chan_ooh323 in function CHANNEL()
2132 libpri channel driver (chan_dahdi) DAHDI changes
2133 --------------------------
2134 * Added moh_signaling option to specify what to do when the channel's bridged
2135 peer puts the ISDN channel on hold.
2136 * Added display_send and display_receive options to control how the display ie
2137 is handled. To send display text from the dialplan use the SendText()
2138 application when the option is enabled.
2139 * Added mcid_send option to allow sending a MCID request on a span.
2142 --------------------------
2143 * Added setvar option to calendar.conf to allow setting channel variables on
2144 notification channels.
2145 * Added "calendar show types" CLI command to list registered calendar
2149 --------------------------
2150 * Added two new options, r and t with file name arguments to record
2151 single direction (unmixed) audio recording separate from the bidirectional
2152 (mixed) recording. The mixed file name argument is optional now as long
2153 as at least one recording option is used.
2156 --------------------------
2157 * Added a new option, l, which will disable local call optimization for
2158 channels involved with the FollowMe thread. Use this option to improve
2159 compatability for a FollowMe call with certain dialplan apps, options, and
2163 --------------------------
2164 * Added option "k" that will automatically close the conference when there's
2165 only one person left when a user exits the conference.
2168 --------------------------
2169 * cel_pgsql now supports the 'extra' column for data added using the
2170 CELGenUserEvent() application.
2173 --------------------------
2174 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2175 in the sample extensions.lua file for syntax details.
2176 * Applications that perform jumps in the dialplan such as Goto will now
2177 execute properly. When pbx_lua detects that the context, extension, or
2178 priority we are executing on has changed it will immediately return control
2179 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2180 the priority after the currently executing priority.
2181 * An autoservice is now started by default for pbx_lua channels. It can be
2182 stopped and restarted using the autoservice_stop() and autoservice_start()
2186 --------------------------
2187 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2188 into a FAXStatus event with an 'Operation' header that will be either
2189 'send', 'receive', and 'gateway'.
2190 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2191 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2192 feature will handle converting a fax call between an audio T.30 fax terminal
2193 and an IFP T.38 fax terminal.
2197 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2198 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2199 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2203 * Added general option negative_penalty_invalid default off. when set
2204 members are seen as invalid/logged out when there penalty is negative.
2205 for realtime members when set remove from queue will set penalty to -1.
2206 * Added queue option autopausedelay when autopause is enabled it will be
2207 delayed for this number of seconds since last successful call if there
2208 was no prior call the agent will be autopaused immediately.
2209 * Added member option ignorebusy this when set and ringinuse is not
2210 will allow per member control of multiple calls as ringinuse does for
2215 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2217 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2218 one participant left (much like a normal call bridge)
2219 * Added extra argument to Originate to set timeout.
2223 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2224 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2225 utility in the UTILS section of menuselect. If an existing astdb is found and no
2226 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2227 convert an existing astdb to the SQLite3 version automatically at runtime.
2231 * Modules marked as deprecated are no longer marked as building by default. Enabling
2232 these modules is still available via menuselect.
2236 * authdebug is now disabled by default. To enable this functionaility again
2237 set authdebug = yes in iax.conf.
2241 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2242 releases it was disabled.
2246 * The PBX core previously made a call with a non-existing extension test for
2247 extension s@default and jump there if the extension existed.
2248 This was a bad default behaviour and violated the principle of least surprise.
2249 It has therefore been changed in this release. It may affect some
2250 applications and configurations that rely on this behaviour. Most channel
2251 drivers have avoided this for many releases by testing whether the extension
2252 called exists before starting the PBX and generating a local error.
2253 This behaviour still exists and works as before.
2255 Extension "s" is used when no extension is given in a channel driver,
2256 like immediate answer in DAHDI or calling to a domain with no user part
2259 ------------------------------------------------------------------------------
2260 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2261 ------------------------------------------------------------------------------
2265 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2266 now defaults to force_rport. It is very important that phones requiring nat=no be
2267 specifically set as such instead of relying on the default setting. If at all
2268 possible, all devices should have nat settings configured in the general section as
2269 opposed to configuring nat per-device.
2270 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2271 codecs sent in response to an INVITE to the single most preferred codec.
2272 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2273 to be used for the outgoing call. It must be one of the codecs configured
2275 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2276 to be used for holding a private key. If tlsprivatekey is not specified,
2277 tlscertfile is searched for both public and private key.
2278 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2279 outbound client connections to be specified.
2280 * The sendrpid parameter has been expanded to include the options
2281 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2282 header to be sent (equivalent to setting sendrpid=yes) and setting
2283 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2284 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2285 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2286 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2287 will accept the SDP even if the SDP version number is not properly incremented,
2288 but will generate a warning in the log indicating that the SIP peer that sent
2289 the SDP should have the 'ignoresdpversion' option set.
2290 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2291 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2292 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2293 remote side requests it and disables symmetric RTP support. Setting it to
2294 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2295 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2296 and enables symmetric RTP support.
2297 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2298 response. This permits the master channel to know how each channel dialled
2299 in a multi-channel setup resolved in an individual way. This carries a
2300 performance penalty and can be disabled in sip.conf using the
2301 'storesipcause' option.
2302 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2303 configuration for the externip and externhost options when tcp or tls is used.
2304 * Added support for message body (stored in content variable) to SIP NOTIFY message
2305 accessible via AMI and CLI.
2306 * Added 'media_address' configuration option which can be used to explicitly specify
2307 the IP address to use in the SDP for media (audio, video, and text) streams.
2308 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2309 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2311 * Added 'use_q850_reason' configuration option for generating and parsing
2312 if available Reason: Q.850;cause=<cause code> header. It is implemented
2313 in some gateways for better passing PRI/SS7 cause codes via SIP.
2314 * When dialing SIP peers, a new component may be added to the end of the dialstring
2315 to indicate that a specific remote IP address or host should be used when dialing
2316 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2317 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2318 ability to selectively force bridged channels to also be encrypted is also
2319 implemented. Branching in the dialplan can be done based on whether or not
2320 a channel has secure media and/or signaling.
2321 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2323 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2324 Charge messages to snom phones.
2325 * Added support for G.719 media streams.
2326 * Added support for 16khz signed linear media streams.
2327 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2328 RTP has been outfitted with the same abilities.
2329 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2330 available in device configurations as well as in the dial plan.
2331 * Addition of the 'subscribe_network_change' option for turning on and off
2332 res_stun_monitor module support in chan_sip.
2333 * Addition of the 'auth_options_requests' option for turning on and off
2334 authentication for OPTIONS requests in chan_sip.
2338 * Add #tryinclude statement for config files. This provides the same
2339 functionality as the #include statement however an asterisk module will
2340 still load if the filename does not exist. Using the #include statement
2341 Asterisk will not allow the module to load.
2345 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2346 on realtime updates.
2347 * Added the ability for chan_iax2 to inform the dialplan whether or not
2348 encryption is being used. This interoperates with the SIP SRTP implementation
2349 so that a secure SIP call can be bridged to a secure IAX call when the
2350 dialplan requires bridged channels to be "secure".
2351 * Addition of the 'subscribe_network_change' option for turning on and off
2352 res_stun_monitor module support in chan_iax.
2357 * Added ability to preset channel variables on indicated lines with the setvar
2358 configuration option. Also, clearvars=all resets the list of variables back
2360 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2361 See configs/res_pktccops.conf for more information.
2363 XMPP Google Talk/Jingle changes
2364 -------------------------------
2365 * Added the externip option to gtalk.conf.
2366 * Added the stunaddr option to gtalk.conf which allows for the automatic
2367 retrieval of the external ip from a stun server.
2371 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2372 match to a partial channel name.
2373 * Added .m3u support for Mp3Player application.
2374 * Added progress option to the app_dial D() option. When progress DTMF is
2375 present, those values are sent immediately upon receiving a PROGRESS message
2376 regardless if the call has been answered or not.
2377 * Added functionality to the app_dial F() option to continue with execution
2378 at the current location when no parameters are provided.
2379 * Added the 'a' option to app_dial to answer the calling channel before any
2380 announcements or macros are executed.
2381 * Modified app_dial to set answertime when the called channel answers even if
2382 the called channel hangs up during playback of an announcement.
2383 * Modified app_dial 'r' option to support an additional parameter to play an
2384 indication tone from indications.conf
2385 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2386 to cycle through the next available channel. By default this is still '*'.
2387 * Added x() option to app_chanspy. This option allows DTMF to be set to
2388 exit the application.
2389 * The Voicemail application has been improved to automatically ignore messages
2390 that only contain silence.
2391 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2392 associated mailbox(es) to be greetings-only.
2393 * The ChanSpy application now has the 'S' option, which makes the application
2394 automatically exit once it hits a point where no more channels are available
2396 * The ChanSpy application also now has the 'E' option, which spies on a single
2397 channel and exits when that channel hangs up.
2398 * The MeetMe application now turns on the DENOISE() function by default, for
2399 each participant. In our tests, this has significantly decreased background
2400 noise (especially noisy data centers).
2401 * Voicemail now permits storage of secrets in a separate file, located in the
2402 spool directory of each individual user. The control for this is located in
2403 the "passwordlocation" option in voicemail.conf. Please see the sample
2404 configuration for more information.
2405 * The ChanIsAvail application now exposes the returned cause code using a separate
2406 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2407 * Added 'd' option to app_followme. This option disables the "Please hold"
2409 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2410 received will terminate recording.
2411 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2412 Previously the folder could only be set per context, but has now been extended
2413 using the imapfolder option.
2414 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2415 * Voicemail now allows the pager date format to be specified separately from the
2417 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2418 to allow joining, leaving, and sending text to group chats.
2419 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2420 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2421 to all paged phones (and optionally excluding the caller's one using the new
2422 option 'n') before the call is bridged.
2423 * The 'f' option to Dial has been augmented to take an optional argument. If no
2424 argument is provided, the 'f' option works as it always has. If an argument is
2425 provided, then the connected party information of all outgoing channels created
2426 during the Dial will be set to the argument passed to the 'f' option.
2427 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2429 * The OSP lookup application adds in/outbound network ID, optional security,
2430 number portability, QoS reporting, destination IP port, custom info and service
2432 * Added new application VMSayName that will play the recorded name of the voicemail
2433 user if it exists, otherwise will play the mailbox number.
2434 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2435 retrieve state for a particular bridge, where <name> is the conference name
2436 * app_directory now allows exiting at any time using the operator or pound key.
2437 * Voicemail now supports setting a locale per-mailbox.
2438 * Two new applications are provided for declining counting phrases in multiple
2439 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2441 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2443 * Voicemail now includes rdnis within msgXXXX.txt file.
2444 * ExternalIVR now supports IPv6 addresses.
2445 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2446 at https://wiki.asterisk.org/wiki/x/oQBB
2447 * ParkedCall and Park can now specify the parking lot to use.
2451 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2452 over SRV records associated with a specific service. From the CLI, type
2453 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2454 details on how these may be used.
2455 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2456 pitch of a channel's tx and rx audio streams.
2457 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2458 setting various connected line and redirecting party information.
2459 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2460 support ISDN subaddressing.
2461 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2462 * For DAHDI channels, the CHANNEL() dialplan function now allows
2463 the dialplan to request changes in the configuration of the active
2464 echo canceller on the channel (if any), for the current call only.
2467 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2469 The possible values are:
2471 on - normal mode (the echo canceller is actually reinitialized)
2473 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2475 voice - voice mode (returns from FAX mode, reverting the changes that
2476 were made when FAX mode was requested)
2477 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2478 and setting variables on the channel which created the current channel.
2479 Administrators should take care to avoid naming conflicts, when multiple
2480 channels are dialled at once, especially when used with the Local channel
2481 construct (which all could set variables on the master channel). Usage
2482 of the HASH() dialplan function, with the key set to the name of the slave
2483 channel, is one approach that will avoid conflicts.
2484 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2486 * func_odbc now allows multiple row results to be retrieved without using
2487 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2488 from the same query by using the name of the function which retrieved the
2489 first row as an argument to ODBC_FETCH().
2490 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2491 dialplan. This function returns the content of the received message.
2492 * Added REPLACE, which searches a given variable name for a set of characters,
2493 then either replaces them with a single character or deletes them.
2494 * Added PASSTHRU, which literally passes the same argument back as its return
2495 value. The intent is to be able to use a literal string argument to
2496 functions that currently require a variable name as an argument.
2497 * HASH-associated variables now can be inherited across channel creation, by
2498 prefixing the name of the hash at assignment with the appropriate number of
2499 underscores, just like variables.
2500 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2501 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2502 whether or not channels that are bridged to the current channel will be
2503 required to have secure signaling and/or media.
2504 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2505 the current channel has secure signaling and/or media.
2506 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2507 "no_media_path" option.
2508 Returns "0" if there is a B channel associated with the call.
2509 Returns "1" if no B channel is associated with the call. The call is either
2510 on hold or is a call waiting call.
2511 * Added option to dialplan function CDR(), the 'f' option
2512 allows for high resolution times for billsec and duration fields.
2513 * FILE() now supports line-mode and writing.
2514 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2515 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2519 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2520 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2521 and is set when a dynamic feature is triggered.
2522 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2523 to dynamically create a new parking lot matching the value this varible is
2525 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2526 features.conf that should be the base for dynamic parkinglots.
2527 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2528 parkinglot should have.
2529 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2530 parkinglot should have.
2531 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2536 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2537 timeout has expired.
2538 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2539 to the caller when an Agent's phone is ringing. This can be used to indicate
2540 to the caller that their call is about to be picked up, which is nice when
2541 one has been on hold for an extened period of time.
2542 * A new config option, penaltymemberslimit, has been added to queues.conf.
2543 When set this option will disregard penalty settings when a queue has too
2545 * A new option, 'I' has been added to both app_queue and app_dial.
2546 By setting this option, Asterisk will not update the caller with
2547 connected line changes or redirecting party changes when they occur.
2548 * A 'relative-periodic-announce' option has been added to queues.conf. When
2549 enabled, this option will cause periodic announce times to be calculated
2550 from the end of announcements rather than from the beginning.
2551 * The autopause option in queues.conf can be passed a new value, "all." The
2552 result is that if a member becomes auto-paused, he will be paused in all
2553 queues for which he is a member, not just the queue that failed to reach
2555 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2556 * The queue logger now allows events to optionally propagate to a file,
2557 even when realtime logging is turned on. Additionally, realtime logging
2558 supports sending the event arguments to 5 individual fields, although it
2559 will fallback to the previous data definition, if the new table layout is
2562 mISDN channel driver (chan_misdn) changes
2563 ----------------------------------------
2564 * Added display_connected parameter to misdn.conf to put a display string
2565 in the CONNECT message containing the connected name and/or number if
2566 the presentation setting permits it.
2567 * Added display_setup parameter to misdn.conf to put a display string
2568 in the SETUP message containing the caller name and/or number if the
2569 presentation setting permits it.
2570 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2571 indicate the dialplan settings are to be obtained from the asterisk
2573 * Made misdn.conf parameter callerid accept the "name" <number> format
2574 used by the rest of the system.
2575 * Made use the nationalprefix and internationalprefix misdn.conf
2576 parameters to prefix any received number from the ISDN link if that
2577 number has the corresponding Type-Of-Number. NOTE: This includes
2578 comparing the incoming call's dialed number against the MSN list.
2579 * Added the following new parameters: unknownprefix, netspecificprefix,
2580 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2581 received number from the ISDN link if that number has the corresponding
2583 * Added new dialplan application misdn_command which permits controlling
2584 the CCBS/CCNR functionality.
2585 * Added new dialplan function mISDN_CC which permits retrieval of various
2586 values from an active call completion record.
2587 * For PTP, you should manually send the COLR of the redirected-to party
2588 for an incomming redirected call if the incoming call could experience
2589 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2590 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2591 if the REDIRECTING(from-num) is not empty.
2592 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2593 option on all of the REDIRECTING statements before dialing the
2594 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2595 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2596 redirecting-to presentation (COLR) when it becomes available.
2597 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2600 thirdparty mISDN enhancements
2601 -----------------------------
2602 mISDN has been modified by Digium, Inc. to greatly expand facility message
2604 * Enhanced COLP support for call diversion and transfer.
2605 * CCBS/CCNR support.
2607 The latest modified mISDN v1.1.x based version is available at:
2608 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2609 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2611 Tagged versions of the modified mISDN code are available under:
2612 http://svn.digium.com/svn/thirdparty/mISDN/tags
2613 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2615 libpri channel driver (chan_dahdi) DAHDI changes
2616 -------------------------------------------
2617 * The channel variable PRIREDIRECTREASON is now just a status variable
2618 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2619 to read and alter the reason.
2620 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2621 redirected-to party for an incomming redirected call if the incoming call
2622 could experience further redirects. Just set the
2623 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2624 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2626 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2627 use the inhibit(i) option on all of the REDIRECTING statements before
2628 dialing the redirected-to party. You still have to set the
2629 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2630 will update the redirecting-to presentation (COLR) when it becomes available.
2631 * Added the ability to ignore calls that are not in a Multiple Subscriber
2632 Number (MSN) list for PTMP CPE interfaces.
2633 * Added dynamic range compression support for dahdi channels. It is
2634 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2635 * Added support for ISDN calling and called subaddress with partial support
2636 for connected line subaddress.
2637 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2638 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2639 to transfer a held call on disconnect similar to an analog phone.
2640 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2641 Will reroute/deflect an outgoing call when receive the message.
2642 Can use the DAHDISendCallreroutingFacility to send the message for the
2644 * Added standard location to add options to chan_dahdi dialing:
2645 Dial(DAHDI/g1[/extension[/options]])
2648 R Reverse charging indication
2649 * Added Reverse Charging Indication (Collect calls) send/receive option.
2650 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2651 Dial(DAHDI/g1/extension/R)
2652 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2653 (requires latest LibPRI)
2654 * Added ability to send/receive keypad digits in the SETUP message.
2655 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2656 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2657 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2658 (requires latest LibPRI)
2659 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2660 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2661 back into the same interface. Tromboned calls happen because of call routing,
2662 call deflection, call forwarding, and call transfer.
2663 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2664 * Added the ability to support call waiting calls. (The SETUP has no B channel
2666 * Added Malicious Call ID (MCID) event to the AMI call event class.
2667 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2669 Asterisk Manager Interface
2670 --------------------------
2671 * The Hangup action now accepts a Cause header which may be used to
2672 set the channel's hangup cause.
2673 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2674 to specify a separate .pem file to hold a private key. By default sslcert
2675 is used to hold both the public and private key.
2676 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2677 for options containing the 'tls' prefix. For example, 'sslenable' is now
2678 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2679 across all .conf files. All affected sample.conf files have been modified to
2680 reflect this change. Previous options such as 'sslenable' still work,
2681 but options with the 'tls' prefix are preferred.
2682 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2683 in a channel. (res_mutestream.so)
2684 * The configuration file manager.conf now supports a channelvars option, which
2685 specifies a list of channel variables to include in each channel-oriented
2687 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2688 and ExtraPriority to allow redirecting the second channel to a different
2689 location than the first.
2690 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2692 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2693 in a MixMonitor recording.
2694 * The 'iax2 show peers' output is now similar to the expected output of
2696 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2698 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2699 AOC-E messages on a channel.
2700 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2701 conform more closely to similar events.
2702 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2704 * Added optional parkinglot variable for park command.
2705 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2706 if CallerIDNum and CallerIDName headers are also present.
2708 Channel Event Logging
2709 ---------------------
2710 * A new interface, CEL, is introduced here. CEL logs single events, much like
2711 the AMI, but it differs from the AMI in that it logs to db backends much
2712 like CDR does; is based on the event subsystem introduced by Russell, and
2713 can share in all its benefits; allows multiple backends to operate like CDR;
2714 is specialized to event data that would be of concern to billing sytems,
2715 like CDR. Backends for logging and accounting calls have been produced,
2716 but a new CDR backend is still in development.
2720 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2721 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2722 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2723 * Multiple files and formats can now be specified in cdr_custom.conf.
2724 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2725 See configs/cdr_syslog.conf.sample for more information.
2726 * A 'sequence' field has been added to CDRs which can be combined with
2727 linkedid or uniqueid to uniquely identify a CDR.
2728 * Handling of billsec and duration field has changed. If your table definition
2729 specifies those fields as float,double or similar they will now be logged with
2730 microsecond accuracy instead of a whole integer.
2732 Calendaring for Asterisk
2733 ------------------------
2734 * A new set of modules were added supporing calendar integration with Asterisk.
2735 Dialplan functions for reading from and writing to calendars are included,
2736 as well as the ability to execute dialplan logic upon calendar event notifications.
2737 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2738 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2739 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2740 2003 support does not support forms-based authentication).
2742 Call Completion Supplementary Services for Asterisk
2743 ---------------------------------------------------
2744 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2745 DAHDI/ISDN supports call completion for the following switch types:
2746 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2747 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2749 Multicast RTP Support
2750 ---------------------
2751 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2752 The channel driver can be used with the Page application to perform multicast RTP
2753 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2754 Type can be either basic or linksys.
2755 Destination is the IP address and port for the RTP packets.
2756 Control address is specific to the linksys type and is used for sending the control
2757 packets unique to them.
2759 Security Events Framework
2760 -------------------------
2761 * Asterisk has a new C API for reporting security events. The module res_security_log
2762 sends these events to the "security" logger level. Currently, AMI is the only
2763 Asterisk component that reports security events. However, SIP support will be
2764 coming soon. For more information on the security events framework, see the
2765 "Asterisk Security Framework" section of the Asterisk wiki at
2766 https://wiki.asterisk.org/wiki/x/wgBQ
2767 * SIP support was added in Asterisk 10
2768 * This API now supports IPv6 addresses
2772 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2773 * A spandsp based fax backend (res_fax_spandsp) has been added.
2774 * The app_fax module has been deprecated in favor of the res_fax module and
2775 the new res_fax_spandsp backend.
2776 * The SendFAX and ReceiveFAX applications now send their log messages to a
2777 'fax' logger level, instead of to the generic logger levels. To see these
2778 messages, the system's logger.conf file will need to direct the 'fax' logger
2779 level to one or more destinations; the logger.conf.sample file includes an
2780 example of how to do this. Note that if the 'fax' logger level is *not*
2781 directed to at least one destination, log messages generated by these
2782 applications will be lost, and that if the 'fax' logger level is directed to
2783 the console, the 'core set verbose' and 'core set debug' CLI commands will
2784 have no effect on whether the messages appear on the console or not.
2788 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2789 Now, in order to enable transmitting silence during record the transmit_silence
2790 option should be used. transmit_silence_during_record remains a valid option, but
2791 defaults to the behavior of the transmit_silence option.
2792 * Addition of the Unit Test Framework API for managing registration and execution
2793 of unit tests with the purpose of verifying the operation of C functions.
2794 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2795 XMPP text messages to the remote JID.
2796 * Modules.conf has a new option - "require" - that marks a module as critical for
2797 the execution of Asterisk.
2798 If one of the required modules fail to load, Asterisk will exit with a return
2800 * An 'X' option has been added to the asterisk application which enables #exec support.
2801 This allows #exec to be used in asterisk.conf.
2802 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2803 * A new lockconfdir option has been added to asterisk.conf to protect the
2804 configuration directory (/etc/asterisk by default) during reloads.
2805 * The parkeddynamic option has been added to features.conf to enable the creation
2806 of dynamic parkinglots.
2807 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2808 the reportalarms config option.
2809 * chan_dahdi supports dialing configuring and dialing by device file name.
2810 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2811 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2812 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2813 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2814 Handy for the above name-based syntax as it does not depend on
2815 initialization order.
2816 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2817 significant increase in performance (about 3X) for installations using this switchtype.
2818 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2819 AIS. For more information, please see the Distributed Device State section of the
2820 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2821 * The addition of G.719 pass-through support.
2822 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2823 during device configuration.
2824 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2825 have less than 3 lines on the LCD.
2826 * Realtime now supports database failover. See the sample extconfig.conf for details.
2827 * The addition of improved translation path building for wideband codecs. Sample
2828 rate changes during translation are now avoided unless absolutely necessary.
2829 * The addition of the res_stun_monitor module for monitoring and reacting to network
2830 changes while behind a NAT.
2831 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2832 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2833 These allow support for any Administration. Default is AT&T values.
2837 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2838 optionally accept a filename, to apply the setting only to the code generated from
2839 that source file when Asterisk was built. However, there are some modules in Asterisk
2840 that are composed of multiple source files, so this did not result in the behavior
2841 that users expected. In this version, 'core set debug' and 'core set verbose'
2842 can optionally accept *module* names instead (with or without the .so extension),
2843 which applies the setting to the entire module specified, regardless of which source
2844 files it was built from.
2845 * New 'manager show settings' command showing the current settings loaded from
2847 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2848 the channel hangup request to all channels.
2849 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2851 ------------------------------------------------------------------------------
2852 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2853 ------------------------------------------------------------------------------
2857 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2858 Snom phones use this for call pickup of extensions that the phone is
2860 * Added support for setting the domain in the URI for caller of an
2861 outbound call by using the SIPFROMDOMAIN channel variable.
2862 * Added a new configuration option "remotesecret" for authentication to
2863 remote services. For backwards compatibility, "secret" still has the
2864 same function as before, but now you can configure both a remote secret and a
2865 local secret for mutual authentication.
2866 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2867 the sound will be played to the target of an attended transfer
2868 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2869 finer control over how many peers Asterisk will qualify and the gap between them
2870 when all peers need to be qualified at the same time.
2871 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2872 (either globally or for a specific peer), chan_sip will treat any SDP data
2873 it receives as new data and update the media stream accordingly. By
2874 default, Asterisk will only modify the media stream if the SDP session
2875 version received is different from the current SDP session version. This
2876 option is required to interoperate with devices that have non-standard SDP
2877 session version implementations (observed with Microsoft OCS). This option
2878 is disabled by default.
2879 * The parsing of register => lines in sip.conf has been modified to allow a port
2880 to be present in the "user" portion. Please see the sip.conf.sample file for more
2882 * Added support for subscribing to MWI on a remote server and making the status available
2883 as a mailbox. Please see the sip.conf.sample file for more information.
2884 * Added a function to remove SIP headers added in the dialplan before the
2885 first INVITE is generated - SIPRemoveHeader()
2886 * Channel variables set with setvar= in a device configuration is now
2887 set both for inbound and outbound calls.
2888 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2892 * Added immediate option to iax.conf
2893 * Added forceencryption option to iax.conf
2894 * Added Encryption and Trunk status to manager command "iaxpeers"
2898 * The configuration file now holds separate sections for devices and lines.
2899 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2904 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2905 support for LibOpenR2. http://www.libopenr2.org/
2906 * The UK option waitfordialtone has been added for use with BT analog
2908 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2909 is used in conjunction with the 'faxdetect' configuration option. When
2910 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2911 switch to the configured faxbuffers policy. For example, to use 6 buffers
2912 and a 'full' buffer policy for a fax transmission, add:
2914 The faxbuffers configuration will be in affect until the call is torn down.
2915 * Added service message support for 4ESS/5ESS switches.
2919 * For DAHDI channels, the CHANNEL() dialplan function now
2920 supports changing the channel's buffer policy (for the current
2921 call only), using this syntax:
2923 exten => s,n,Set(CHANNEL(buffers)=6,full)
2925 This would change the channel to the 'full' buffer policy and
2926 6 (six) buffers. Possible options for this setting are the same
2927 as those in chan_dahdi.conf.
2928 * Added a new dialplan function, CURLOPT, which permits setting various
2929 options that may be useful with the CURL dialplan function, such as
2930 cookies, proxies, connection timeouts, passwords, etc.
2931 * Permit the syntax and synopsis fields of the corresponding dialplan
2932 functions to be individually set from func_odbc.conf.
2933 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2934 * func_odbc now may specify an insert query to execute, when the write query
2935 affects 0 rows (usually indicating that no such row exists).
2936 * Added a new dialplan function, LISTFILTER, which permits removing elements
2937 from a set list, by name. Uses the same general syntax as the existing CUT
2938 and FIELDQTY dialplan functions, which also manage lists.
2939 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2940 obtaining realtime data from the dialplan.
2941 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2942 a subroutine when using the GoSub() and Return() applications.
2943 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2944 of "core show function AUDIOHOOK_INHERIT" from the CLI
2945 * Added AES_ENCRYPT. For information on its use, please see the output
2946 of "core show function AES_ENCRYPT" from the CLI
2947 * Added AES_DECRYPT. For information on its use, please see the output
2948 of "core show function AES_DECRYPT" from the CLI
2949 * func_odbc now supports database transactions across multiple queries.
2953 * Scheduled meetme conferences may now have their end times extended by
2955 * app_authenticate now gives the ability to select a prompt other than
2957 * app_directory now pays attention to the searchcontexts setting in
2958 voicemail.conf and will look through all contexts, if no context is
2959 specified in the initial argument.
2960 * A new application, Originate, has been introduced, that allows asynchronous
2961 call origination from the dialplan.
2962 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2963 in addition to the setting in the "general" context.
2964 * Added ConfBridge dialplan application which does conference bridges without
2965 DAHDI. For information on its use, please see the output of
2966 "core show application ConfBridge" from the CLI.
2970 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2971 operation to the AMI Redirect action.
2972 * extensions.conf now allows you to use keyword "same" to define an extension
2973 without actually specifying an extension. It uses exactly the same pattern
2974 as previously used on the last "exten" line. For example:
2975 exten => 123,1,NoOp(something)
2976 same => n,SomethingElse()
2977 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2978 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2979 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2980 by the new clialiases module. See cli_aliases.conf.sample file.
2981 * Times within timespecs are now accurate down to the minute. This is a change
2982 from historical Asterisk, which only provided timespecs rounded to the nearest
2983 even (read: evenly divisible by 2) minute mark.
2984 * The realtime switch now supports an option flag, 'p', which disables searches for
2986 * In addition to a time range and date range, timespecs now accept a 5th optional
2987 argument, timezone. This allows you to perform time checks on alternate
2988 timezones, especially if those daylight savings time ranges vary from your
2989 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2991 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2992 give you the correct output for an asterisk box behind nat. It will give you the
2993 externhost and localnet settings.
2994 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2995 can connect calls in passthrough mode, as well as record and play back files.
2996 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2997 using pickupsound and pickupfailsound in features.conf.
2998 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2999 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3000 instead of the /var/run/asterisk.pid where it used to be. This will make
3001 installs as non-root easier to manage.
3006 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3007 be written; they will no longer be explicitly written.
3009 Asterisk Manager Interface
3010 --------------------------
3011 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3012 a non-empty value) in your request. If you do this, any pending AMI events will
3013 *not* be included in the response to your request as they would normally, but
3014 will be left in the event queue for the next request you make to retrieve. For
3015 some applications, this will allow you to guarantee that you will only see
3016 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3017 To know whether the Asterisk server supports this header or not, your client can
3018 inspect the first response back from the server to see if it includes this header:
3020 Pragma: SuppressEvents
3022 If this is included, the server supports event suppression.
3024 * Added 4 new Actions to list skinny device(s) and line(s)
3030 LDAP Schema File Additions
3031 --------------------------
3032 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3033 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3035 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3036 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3037 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3038 * Removed redundant IPaddr (there's already IPAddress)
3039 - Gives more configuration Flags for SIP-Users available (tested)
3040 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3041 without extensibleObject (which really should be the last resort); gives
3042 also additional possibilities for LDAP-filter
3044 ------------------------------------------------------------------------------
3045 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3046 ------------------------------------------------------------------------------
3048 Device State Handling
3049 ---------------------
3050 * The event infrastructure in Asterisk got another big update to help support
3051 distributed events. It currently supports distributed device state and
3052 distributed Voicemail MWI (Message Waiting Indication). A new module has
3053 been merged, res_ais, which facilitates communicating events between servers.
3054 It uses the SAForum AIS (Service Availability Forum Application Interface
3055 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3056 a cluster of Asterisk servers, and to share events between them. For more
3057 information on setting this up, refer to the Distributed Device State section
3058 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3062 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3063 variables from an Asterisk configuration file.
3064 * The JACK_HOOK function now has a c() option to supply a custom client name.
3065 * Added two new dialplan functions from libspeex for audio gain control and
3066 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3067 rx directions of a channel from the dialplan.
3068 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3069 based on other parameters. The default is still to search based on the
3070 forwarding station ID. However, there are new options that allow you to search
3071 based on the message desk terminal ID, or the message desk number.
3072 * TIMEOUT() has been modified to be accurate down to the millisecond.
3073 * ENUM*() functions now include the following new options:
3074 - 'u' returns the full URI and does not strip off the URI-scheme.
3075 - 's' triggers ISN specific rewriting
3076 - 'i' looks for branches into an Infrastructure ENUM tree
3077 - 'd' for a direct DNS lookup without any flipping of digits.
3078 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3079 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3080 deviation of jitter, rtt, and loss for a call using chan_sip.
3082 DAHDI channel driver (chan_dahdi) Changes
3083 ----------------------------------------
3084 * Channels can now be configured using named sections in chan_dahdi.conf, just
3085 like other channel drivers, including the use of templates.
3086 * The default for pridialplan has changed from 'national' to 'unknown'.
3090 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3091 to something that matches the pattern a hint will be created using the contents
3092 and variables evaluated.
3093 * Dialplan matching has been extended to allow an extension to return to the
3094 PBX core to wait for more digits. This is done by using the new dialplan
3095 application called "Incomplete". This will permit a whole new level of
3096 extension control, by giving the administrator more control over early
3097 matches employing one of the short-circuit pattern match operators. Note
3098 that custom applications can trigger this same behavior by returning the
3099 special value AST_PBX_INCOMPLETE.
3103 * Directory now permits both first and last names to be matched at the same
3104 time. In addition, the number of digits to enter of the name can be set in
3105 the arguments to Directory; previously, you could enter only 3, regardless
3106 of how many names are in your company. For large companies, this should be
3108 * Voicemail now permits a mailbox setting to wrap around from first to last
3109 messages, if the "messagewrap" option is set to a true value.
3110 * Voicemail now permits an external script to be run, for password validation.
3111 The script should output "VALID" or "INVALID" on stdout, depending upon the
3112 wish to validate or invalidate the password given. Arguments are:
3113 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3115 * Dial has a new option: F(context^extension^pri), which permits a callee to
3116 continue in the dialplan, at the specified label, if the caller hangs up.
3117 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3118 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3119 * The Jack application now has a c() option to supply a custom client name.
3120 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3121 like the pre-existing whisper mode, except that the spy can also talk to the
3122 participant on the bridged channel as well.
3123 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3124 to be spoken instead of the channel name or number. For more information on the
3125 use of this option, issue the command "core show application ChanSpy" from the
3127 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3128 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3129 words, if using the 'd' option, it is not possible to enter a number to append to
3130 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3131 change to whisper mode, and pressing 6 will change to barge mode.
3132 * ExternalIVR now takes several options that affect the way it performs, as
3133 well as having several new commands. Please see the External IVR page on the Asterisk
3134 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3135 * Added ability to communicate over a TCP socket instead of forking a child process for the
3136 ExternalIVR application.
3137 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3138 of just the first one if you give the function more then one channel to check.
3139 * PrivacyManager now takes an option where you can specify a context where the
3140 given number will be matched. This way you have more control over who is allowed
3141 and it stops the people who blindly enter 10 digits.
3142 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3143 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3144 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3145 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3146 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3147 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3148 * The Dial() application no longer copies the language used by the caller to the callee's
3149 channel. If you desire for the caller's channel's language to be used for file playback
3150 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3151 * SendImage() no longer hangs up the channel on error; instead, it sets the
3152 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3153 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3155 * Park has a new option, 's', which silences the announcement of the parking space number.
3156 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3157 invalid input and will be assumed to mean that no timeout is desired.
3161 * Added DNS manager support to registrations for peers referencing peer entries.
3162 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3163 as well as periodically updating the IP address. These properties allow for
3164 better performance as well as recovery in the event of an IP change.
3165 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3166 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3167 These changes also provide performance improvements for call setup and tear down.
3168 * Added ability to specify registration expiry time on a per registration basis in
3170 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3172 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3173 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3174 * 'sip show peers' and 'sip show users' display their entries sorted in
3175 alphabetical order, as opposed to the order they were in, in the config
3177 * Videosupport now supports an additional option, "always", which always sets
3178 up video RTP ports, even on clients that don't support it. This helps with
3179 callfiles and certain transfers to ensure that if two video phones are
3180 connected, they will always share video feeds.
3184 * Existing DNS manager lookups extended to check for SRV records.
3185 * IAX2 encryption support has been improved to support periodic key rotation
3186 within a call for enhanced security. The option "keyrotate" has been
3187 provided to disable this functionality to preserve backwards compatibility
3188 with older versions of IAX2 that do not support key rotation.
3192 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3193 data tree based on the given <path>.
3194 * New CLI command "data show providers" that will display all the registered
3196 * New CLI command, "config reload <file.conf>" which reloads any module that
3197 references that particular configuration file. Also added "config list"
3198 which shows which configuration files are in use.
3199 * New CLI commands, "pri show version" and "ss7 show version" that will
3200 display which version of libpri and libss7 are being used, respectively.
3201 A new API call was added so trunk will now have to be compiled against
3202 a versions of libpri and libss7 that have them or it will not know that
3203 these libraries exist.
3204 * The commands "core show globals", "core set global" and "core set chanvar" has
3205 been deprecated in favor of the more semanticly correct "dialplan show globals",
3206 "dialplan set chanvar" and "dialplan set global".
3207 * New CLI command "dialplan show chanvar" to list all variables associated
3208 with a given channel.
3212 * Addresses managed by DNS manager now can check to see if there is a DNS
3213 SRV record for a given domain and will use that hostname/port if present.
3215 AMI - The manager (TCP/TLS/HTTP)
3216 --------------------------------
3217 * The Status command now takes an optional list of variables to display
3218 along with channel status.
3219 * The QueueEntry event now also includes the channel's uniqueid
3223 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3224 as some people were running into this limit. This limit has been increased
3229 * The TRANSFER queue log entry now includes the the caller's original
3230 position in the transferred-from queue.
3231 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3232 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3233 as well as an explanation about timeout options in general
3234 * Added a new option - C - for forcing the "answered elsewhere" flag on
3235 cancellation of calls in to members of the queue. This is to avoid the
3236 call to a member of a queue having the call listed as a "missed call".
3240 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3241 adaptive capabilities. What this means in practical terms is that if your
3242 realtime table lacks critical fields, Asterisk will now emit warnings to
3243 that effect. Also, some of the realtime drivers have the ability (if
3244 configured) to automatically add those columns to the table with the
3245 correct type and length.
3249 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3250 the 'setvar' option to cause a given audio file to be played upon completion
3251 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3252 Skinny channels only.
3253 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3254 Hoard page on the Asterisk wiki for more information:
3255 https://wiki.asterisk.org/wiki/x/pQBB
3256 * Config file variables may now be appended to, by using the '+=' append
3257 operator. This is most helpful when working with long SQL queries in
3258 func_odbc.conf, as the queries no longer need to be specified on a single
3260 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3261 which will add a second to the billsec when the ending
3262 time is set, if the number in the microseconds field of the end time is
3263 greater than the number of microseconds in the answer time. This allows
3264 users to count the 'initiated' seconds in their billing records.
3266 ------------------------------------------------------------------------------
3267 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3268 ------------------------------------------------------------------------------
3270 AMI - The manager (TCP/TLS/HTTP)
3271 --------------------------------
3272 * Manager has undergone a lot of changes, all of them documented
3273 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3274 * Manager version has changed to 1.1
3275 * Added a new action 'CoreShowChannels' to list currently defined channels
3276 and some information about them.
3277 * Added a new action 'SIPshowregistry' to list SIP registrations.
3278 * Added TLS support for the manager interface and HTTP server
3279 * Added the URI redirect option for the built-in HTTP server
3280 * The output of CallerID in Manager events is now more consistent.
3281 CallerIDNum is used for number and CallerIDName for name.
3282 * Enable https support for builtin web server.
3283 See configs/http.conf.sample for details.
3284 * Added a new action, GetConfigJSON, which can return the contents of an
3285 Asterisk configuration file in JSON format. This is intended to help
3286 improve the performance of AJAX applications using the manager interface
3288 * SIP and IAX manager events now use "ChannelType" in all cases where we
3289 indicate channel driver. Previously, we used a mixture of "Channel"
3290 and "ChannelDriver" headers.
3291 * Added a "Bridge" action which allows you to bridge any two channels that
3292 are currently active on the system.
3293 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3294 the voicemail users setup.
3295 * Added 'DBDel' and 'DBDelTree' manager commands.
3296 * cdr_manager now reports events via the "cdr" level, separating it from
3297 the very verbose "call" level.
3298 * Manager users are now stored in memory. If you change the manager account
3299 list (delete or add accounts) you need to reload manager.
3300 * Added Masquerade manager event for when a masquerade happens between
3302 * Added "manager reload" command for the CLI
3303 * Lots of commands that only provided information are now allowed under the
3304 Reporting privilege, instead of only under Call or System.
3305 * The IAX* commands now require either System or Reporting privilege, to
3306 mirror the privileges of the SIP* commands.
3307 * Added ability to retrieve list of categories in a config file.
3308 * Added ability to retrieve the content of a particular category.
3309 * Added ability to empty a context.
3310 * Created new action to create a new file.
3311 * Updated delete action to allow deletion by line number with respect to category.
3312 * Added new action insert to add new variable to category at specified line.
3313 * Updated action newcat to allow new category to be inserted in file above another
3315 * Added new event "JitterBufStats" in the IAX2 channel
3316 * Originate now requires the Originate privilege and, if you want to call out
3317 to a subshell, it requires the System privilege, as well. This was done to
3318 enhance manager security.
3319 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3320 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3321 or manager show command Atxfer from the CLI
3322 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3323 details or manager show command IAXregistry from the CLI
3327 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3328 state in the dialplan, as well as creating custom device states that are
3329 controllable from the dialplan.
3330 * Extend CALLERID() function with "pres" and "ton" parameters to
3331 fetch string representation of calling number presentation indicator
3332 and numeric representation of type of calling number value.
3333 * MailboxExists converted to dialplan function
3334 * A new option to Dial() for telling IP phones not to count the call
3335 as "missed" when dial times out and cancels.
3336 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3337 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3338 held for any given channel. Also, locks are automatically freed when a
3340 * Added HINT() dialplan function that allows retrieving hint information.
3341 Hints are mappings between extensions and devices for the sake of
3342 determining the state of an extension. This function can retrieve the list
3343 of devices or the name associated with a hint.
3344 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3346 * Added SYSINFO() dialplan function which allows retrieval of system information
3347 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3348 the existence of a dialplan target.
3349 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3350 upper and lower case, respectively.
3351 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3352 ID for the call (not the Asterisk call ID or unique ID), provided that the
3353 channel driver supports this. For SIP, you get the SIP call-ID for the
3354 bridged channel which you can store in the CDR with a custom field.
3358 * Added CLI permissions, config file: cli_permissions.conf
3359 default is to allow all commands for every local user/group.
3360 Also this new feature added three new CLI commands:
3361 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3362 - cli reload permissions
3363 - cli show permissions
3364 * New CLI command "core show hint" (usage: core show hint <exten>)
3365 * New CLI command "core show settings"
3366 * Added 'core show channels count' CLI command.
3367 * Added the ability to set the core debug and verbose values on a per-file basis.
3368 * Added 'queue pause member' and 'queue unpause member' CLI commands
3369 * Ability to set process limits ("ulimit") without restarting Asterisk
3370 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3371 output to make debugging on busy systems much easier.
3372 * New CLI commands "dialplan set extenpatternmatching true/false"
3373 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3374 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3375 listed in the startup_commands section of cli.conf will get executed.
3376 * Added a CLI command, "devstate change", which allows you to set custom device
3377 states from the func_devstate module that provides the DEVICE_STATE() function
3378 and handling of the "Custom:" devices.
3379 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3380 sorted into the different possible callbacks, with the number of entries
3381 currently scheduled for each. Gives you a feel for how busy the sip channel
3383 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3384 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3385 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3389 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3390 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3391 for a received call. If it is detected, the channel will jump to the
3392 'fax' extension in the dialplan.
3393 * The default SIP useragent= identifier now includes the Asterisk version
3394 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3395 If set, and the incoming request carries authentication info,
3396 the username to match in the users list is taken from the Digest header
3397 rather than from the From: field. This feature is considered experimental.
3398 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3399 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3400 * The "localmask" setting was removed in version 1.2 and the reminder about it
3401 being removed is now also removed.
3402 * A new option "busylevel" for setting a level of calls where asterisk reports
3403 a device as busy, to separate it from call-limit. This value is also added
3404 to the SIP_PEER dialplan function.
3405 * A new realtime family called "sipregs" is now supported to store SIP registration
3406 data. If this family is defined, "sippeers" will be used for configuration and
3407 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3408 registration data, as before.
3409 * The SIPPEER function have new options for port address, call and pickup groups
3410 * Added support for T.140 realtime text in SIP/RTP
3411 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3412 required due to the restructuring of how MWI is handled. See the descriptions
3413 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3414 for more information.
3415 * Added rtpdest option to CHANNEL() dialplan function.
3416 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3417 * SIP now adds a header to the CANCEL if the call was answered by another phone
3418 in the same dial command, or if the new c option in dial() is used.
3419 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3420 states it is not needed. For phones, however, that do require it the "registertrying" option
3421 has been added so it can be enabled.
3422 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3423 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3424 used to enable this functionality).
3425 * New settings for timer T1 and timer B on a global level or per device. This makes it
3426 possible to force timeout faster on non-responsive SIP servers. These settings are
3427 considered advanced, so don't use them unless you have a problem.
3428 * Added a dial string option to be able to set the To: header in an INVITE to any
3430 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3431 the qualify frequency.
3432 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3433 were not properly torn down due to network or endpoint failures during an established
3435 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3436 and configs/sip.conf.sample for more information on how it is used.
3437 * Added a new configuration option "authfailureevents" that enables manager events when
3438 a peer can't authenticate properly.
3439 * Added DNS manager support to registrations for peers not referencing a peer entry.
3443 * Added the trunkmaxsize configuration option to chan_iax2.
3444 * Added the srvlookup option to iax.conf
3445 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3448 XMPP Google Talk/Jingle changes
3449 -------------------------------
3450 * Added the bindaddr option to gtalk.conf.
3454 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
projects / asterisk/asterisk.git / blob