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 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
21 conference user menus.
23 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
24 menus, bridge settings, and user settings that have been applied by the
25 CONFBRIDGE dialplan function.
27 * The ConfBridge dialplan application now sets a channel variable,
28 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
29 how a channel exited the conference.
31 * Added conference user option 'announce_join_leave_review'. This option
32 implies 'announce_join_leave' with the added effect that the user will
33 be asked if they want to confirm or re-record the recording of their
34 name when entering the conference
37 --------------------------
38 * At exit, the Directory application now sets a channel variable
39 DIRECTORY_RESULT to one of the following based on the reason for exiting:
40 OPERATOR user requested operator by pressing '0' for operator
41 ASSISTANT user requested assistant by pressing '*' for assistant
42 TIMEOUT user pressed nothing and Directory stopped waiting
43 HANGUP user's channel hung up
44 SELECTED user selected a user from the directory and is routed
45 USEREXIT user pressed '#' from the selection prompt to exit
46 FAILED directory failed in a way that wasn't accounted for. Dang.
49 --------------------------
50 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
51 and for the channel executing Page respectively.
54 --------------------------
55 * PickupChan now accepts channel uniqueids of channels to pickup.
58 --------------------------
59 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
60 to 'true' (case insensitive), then any Say application (SayNumber,
61 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
62 anticipate DTMF. If DTMF is received, these applications will behave like
63 the background application and jump to the received extension once a match
64 is established or after a short period of inactivity.
67 -------------------------
68 * A new function, MIXMONITOR, has been added to allow access to individual
69 instances of MixMonitor on a channel.
71 ------------------------------------------------------------------------------
72 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
73 ------------------------------------------------------------------------------
78 Asterisk 12 is a standard release of the Asterisk project. As such, the
79 focus of development for this release was on core architectural changes and
80 major new features. This includes:
81 * A more flexible bridging core based on the Bridging API
82 * A new internal message bus, Stasis
83 * Major standardization and consistency improvements to AMI
84 * Addition of the Asterisk RESTful Interface (ARI)
85 * A new SIP channel driver, chan_pjsip
86 In addition, as the vast majority of bridging in Asterisk was migrated to the
87 Bridging API used by ConfBridge, major changes were made to most of the
88 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
90 Specifications have been written for the affected interfaces. These
91 specifications are available on the Asterisk wiki:
92 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
93 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
94 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
96 It is *highly* recommended that anyone migrating to Asterisk 12 read the
97 information regarding its release both in this file and in the accompanying
98 UPGRADE.txt file. More detailed information on the major changes can be found
99 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
104 * Added build option DISABLE_INLINE. This option can be used to work around a
105 bug in gcc. For more information, see
106 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
108 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
109 the CHANNEL_TRACE build option were incompatible with the new bridging
112 * Asterisk now optionally uses libxslt to improve XML documentation generation
113 and maintainability. If libxslt is not available on the system, some XML
114 documentation will be incomplete.
116 * Asterisk now depends on libjansson. If a package of libjansson is not
117 available on your distro, please see http://www.digip.org/jansson/.
119 * Asterisk now depends on libuuid and, optionally, uriparser. It is
120 recommended that you install uriparser, even if it is optional.
122 * The new SIP stack and channel driver uses a particular version of PJSIP.
123 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
124 configuring and installing PJSIP for usage with Asterisk.
126 * Optional API was re-implemented to be more portable, and no longer requires
127 weak reference support from the compiler. The build option OPTIONAL_API may
128 be disabled to disable Optional API support.
135 * Along with AgentRequest, this application has been modified to be a
136 replacement for chan_agent. The act of a channel calling the AgentLogin
137 application places the channel into a pool of agents that can be
138 requested by the AgentRequest application. Note that this application, as
139 well as all other agent related functionality, is now provided by the
140 app_agent_pool module. See chan_agent and AgentRequest for more information.
142 * This application no longer performs agent authentication. If authentication
143 is desired, the dialplan needs to perform this function using the
144 Authenticate or VMAuthenticate application or through an AGI script before
147 * If this application is called and the agent is already logged in, the
148 dialplan will continue exection with the AGENT_STATUS channel variable set
149 to ALREADY_LOGGED_IN.
151 * The agents.conf schema has changed. Rather than specifying agents on a
152 single line in comma delineated fashion, each agent is defined in a separate
153 context. This allows agents to use the power of context templates in their
156 * A number of parameters from agents.conf have been removed. This includes
157 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
158 urlprefix, and savecallsin. These options were obsoleted by the move from
159 a channel driver model to the bridging/application model provided by
164 * A new application, this will request a logged in agent from the pool and
165 bridge the requested channel with the channel calling this application.
166 Logged in agents are those channels that called the AgentLogin application.
167 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
168 application will be set with an appropriate error value.
172 * This application has been removed. It was a holdover from when
173 AgentCallbackLogin was removed.
177 * Added support for additional Ademco DTMF signalling formats, including
178 Express 4+1, Express 4+2, High Speed and Super Fast.
180 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
181 call time, in milliseconds, to run the application.
183 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
184 maximum number of times to retry the call.
186 * Added a new configuration option answait. If set, the AlarmReceiver
187 application will wait the number of milliseconds specified by answait
188 after the channel has answered. Valid values range between 500
189 milliseconds and 10000 milliseconds.
191 * Added configuration option no_group_meta. If enabled, grouping of metadata
192 information in the AlarmReceiver log file will be skipped.
196 * A new application in Asterisk, this will place the calling channel
197 into a holding bridge, optionally entertaining them with some form of
198 media. Channels participating in a holding bridge do not interact with
199 other channels in the same holding bridge. Optionally, however, a channel
200 may join as an announcer. Any media passed from an announcer channel is
201 played to all channels in the holding bridge. Channels leave a holding
202 bridge either when an optional timer expires, or via the ChannelRedirect
203 application or AMI Redirect action.
207 * All participants in a bridge can now be kicked out of a conference room
208 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
209 command, i.e., 'confbridge kick <conference> all'
211 * CLI output for the 'confbridge list' command has been improved. When
212 displaying information about a particular bridge, flags will now be shown
213 for the participating users indicating properties of that user.
215 * The ConfbridgeList event now contains the following fields: WaitMarked,
216 EndMarked, and Waiting. This displays additional properties about the
217 user's profile, as well as whether or not the user is waiting for a
218 Marked user to enter the conference.
220 * Added a new option for conference recording, record_file_append. If enabled,
221 when the recording is stopped and then re-started, the existing recording
222 will be used and appended to.
224 * ConfBridge now has the ability to set the language of announcements to the
225 conference. The language can be set on a bridge profile in confbridge.conf
226 or by the dialplan function CONFBRIDGE(bridge,language)=en.
230 * The channel variable CPLAYBACKSTATUS may now return the value
231 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
232 such as AMI. See the AMI action ControlPlayback for more information.
236 * Added the 'a' option, which allows the caller to enter in an additional
237 alias for the user in the directory. This option must be used in conjunction
238 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
239 specified in voicemail.conf.
243 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
244 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
245 containing the unique ID of the bridge that the channel happens to be in.
249 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
250 for more information.
252 * Variables are no longer purged from the original CDR. See the 'v' option for
255 * The 'A' option has been removed. The Answer time on a CDR is never updated
258 * The 'd' option has been removed. The disposition on a CDR is a function of
259 the state of the channel and cannot be altered.
261 * The 'D' option has been removed. Who the Party B is on a CDR is a function
262 of the state of the respective channels involved in the CDR and cannot be
265 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
266 such that the start time and, if applicable, the answer time was updated.
267 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
268 'r' option now triggers the Reset, setting the start time (and answer time
269 if applicable) to the current time. Note that the 'a' option still sets
270 the answer time to the current time if the channel was already answered.
272 * The 's' option has been removed. A variable can be set on the original CDR
273 if desired using the CDR function, and removed from a forked CDR using the
276 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
277 longer applies in the CDR engine.
279 * The 'v' option now prevents the copy of the variables from the original CDR
280 to the forked CDR. Previously the variables were always copied but were
281 removed from the original. This was changed as removing variables from a CDR
282 can have unintended side effects - this option allows the user to prevent
283 propagation of variables from the original to the forked without modifying
288 * Added the 'n' option to MeetMe to prevent application of the DENOISE
289 function to a channel joining a conference. Some channel drivers that vary
290 the number of audio samples in a voice frame will experience significant
291 quality problems if a denoiser is attached to the channel; this option gives
292 them the ability to remove the denoiser without having to unload func_speex.
296 * The 'b' option now includes conferences as well as sounds played to the
299 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
300 running during a transfer. If a MixMonitor is started on a channel,
301 the MixMonitor will continue to record the audio passing through the
302 channel even in the presence of transfers.
306 * The NoCDR application is deprecated. Please use the CDR_PROP function to
309 * While the NoCDR application will prevent CDRs for a channel from being
310 propagated to registered CDR backends, it will not prevent that data from
311 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
312 function that enables CDRs on a channel will restore those records that have
313 not yet been finalized.
317 * The app_parkandannounce module has been removed. The application
318 ParkAndAnnounce is now provided by the res_parking module. See the
319 res_parking changes for more information.
323 * Added queue available hint. The hint can be added to the dialplan using the
324 following syntax: exten,hint,Queue:{queue_name}_avail
325 For example, if the name of the queue is 'markq':
326 exten => 8501,hint,Queue:markq_avail
327 This will report 'InUse' if there are no logged in agents or no free agents.
328 It will report 'Idle' when an agent is free.
330 * Queues now support a hint for member paused state. The hint uses the form
331 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
332 are the name of the queue and the name of the member to subscribe to,
333 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
334 Members will show as In Use when paused.
336 * The configuration options eventwhencalled and eventmemberstatus have been
337 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
338 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
339 sent. The "Variable" fields will also no longer exist on the Agent* events.
340 These events can be filtered out from a connected AMI client using the
341 eventfilter setting in manager.conf.
343 * The queue log now differentiates between blind and attended transfers. A
344 blind transfer will result in a BLINDTRANSFER message with the destination
345 context and extension. An attended transfer will result in an
346 ATTENDEDTRANSFER message. This message will indicate the method by which
347 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
348 for running an application on a bridge or channel, or "LINK" for linking
349 two bridges together with local channels. The queue log will also now detect
350 externally initiated blind and attended transfers and record the transfer
353 * When performing queue pause/unpause on an interface without specifying an
354 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
355 least one member of any queue exists for that interface.
357 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
358 for realtime queue log entries.
362 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
363 CDRs when they were previously disabled on a channel.
365 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
366 backends occurs on an as-needed basis in order to preserve linkedid
367 propagation and other needed behavior.
371 * A new application, this is similar to SayAlpha except that it supports
372 case sensitive playback of the specified characters. For example,
373 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
377 * This application is deprecated in favor of CHANNEL(amaflags).
381 * The SendDTMF application will now accept 'W' as valid input. This will cause
382 the application to delay one second while streaming DTMF.
386 * A new application in Asterisk 12, this hands control of the channel calling
387 the application over to an external system. Currently, external systems
388 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
392 * UserEvent will now handle duplicate keys by overwriting the previous value
395 * In addition to AMI, UserEvent invocations will now be distributed to any
396 interested Stasis applications.
400 * The voicemail.conf configuration file now has an 'alias' configuration
401 parameter for use with the Directory application. The voicemail realtime
402 database table schema has also been updated with an 'alias' column.
407 * Pass through support has been added for both VP8 and Opus.
409 * Added format attribute negotiation for the Opus codec. Format attribute
410 negotiation is provided by the res_format_attr_opus module.
415 * Masquerades as an operation inside Asterisk have been effectively hidden
416 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
417 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
418 dropping of frame/audio hooks, and other internal implementation details
419 that users had to deal with. This fundamental change has large implications
420 throughout the changes documented for this version. For more information
421 about the new core architecture of Asterisk, please see the Asterisk wiki.
423 * Multiple parties in a bridge may now be transferred. If a participant in a
424 multi-party bridge initiates a blind transfer, a Local channel will be used
425 to execute the dialplan location that the transferer sent the parties to. If
426 a participant in a multi-party bridge initiates an attended transfer,
427 several options are possible. If the attended transfer results in a transfer
428 to an application, a Local channel is used. If the attended transfer results
429 in a transfer to another channel, the resulting channels will be merged into
432 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
433 driver specific. If the channel variable is set on the transferrer channel,
434 the sound will be played to the target of an attended transfer.
436 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
437 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
438 listed. Any more peers in the bridge will not be included in the list.
439 BRIDGEPEER is not valid in holding bridges like parking since those channels
440 do not talk to each other even though they are in a bridge.
442 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
443 and will contain a value if the BRIDGEPEER's channel driver supports it.
445 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
446 was responsible for an attended transfer in a similar fashion to
449 * Modules using the Configuration Framework or Sorcery must have XML
450 configuration documentation. This configuration documentation is included
451 with the rest of Asterisk's XML documentation, and is accessible via CLI
452 commands. See the CLI changes for more information.
454 AMI (Asterisk Manager Interface)
456 * Major changes were made to both the syntax as well as the semantics of the
457 AMI protocol. In particular, AMI events have been substantially improved
458 in this version of Asterisk. For more information, please see the AMI
459 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
461 * AMI events that reference a particular channel or bridge will now always
462 contain a standard set of fields. When multiple channels or bridges are
463 referenced in an event, fields for at least some subset of the channels
464 and bridges in the event will be prefixed with a descriptive name to avoid
465 name collisions. See the AMI event documentation on the Asterisk wiki for
468 * The CLI command 'manager show commands' no longer truncates command names
469 longer than 15 characters and no longer shows authorization requirement
470 for commands. 'manager show command' now displays the privileges needed
471 for using a given manager command instead.
473 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
474 peer in its response if the peer has a subscribe context set.
476 * The SIPqualifypeer action now acknowledges the request once it has
477 established that the request is against a known peer. It also issues a new
478 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
480 * The PlayDTMF action now supports an optional 'Duration' parameter. This
481 specifies the duration of the digit to be played, in milliseconds.
483 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
484 updates when changes occur instead of requiring the use of pollmailboxes.
486 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
487 AMI client to manipulate audio currently being played back on a channel. The
488 supported operations depend on the application being used to send audio to
489 the channel. When the audio playback was initiated using the ControlPlayback
490 application or CONTROL STREAM FILE AGI command, the audio can be paused,
491 stopped, restarted, reversed, or skipped forward. When initiated by other
492 mechanisms (such as the Playback application), the audio can be stopped,
493 reversed, or skipped forward.
495 * Channel related events now contain a snapshot of channel state, adding new
496 fields to many of these events.
498 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
499 in a future release. Please use the common 'Exten' field instead.
501 * The AMI event 'UserEvent' from app_userevent now contains the channel state
502 fields. The channel state fields will come before the body fields.
504 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
505 'UnParkedCall' have changed significantly in the new res_parking module.
507 The 'Channel' and 'From' headers are gone. For the channel that was parked
508 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
509 has a number of fields associated with it. The old 'Channel' header relayed
510 the same data as the new 'ParkeeChannel' header.
512 The 'From' field was ambiguous and changed meaning depending on the event.
513 for most of these, it was the name of the channel that parked the call
514 (the 'Parker'). There is no longer a header that provides this channel name,
515 however the 'ParkerDialString' will contain a dialstring to redial the
516 device that parked the call.
518 On UnParkedCall events, the 'From' header would instead represent the
519 channel responsible for retrieving the parkee. It receives a channel
520 snapshot labeled 'Retriever'. The 'from' field is is replaced with
523 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
525 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
526 fashion has changed the field names 'StartExten' and 'StopExten' to
527 'StartSpace' and 'StopSpace' respectively.
529 * The deprecated use of | (pipe) as a separator in the channelvars setting in
530 manager.conf has been removed.
532 * Channel Variables conveyed with a channel no longer contain the name of the
533 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
534 ChanVariable: bar=baz. When multiple channels are present in a single AMI
535 event, the various ChanVariable fields will contain a suffix that specifies
536 which channel they correspond to.
538 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
539 event always conveys the AMI event for a particular channel.
541 * All 'Reload' events have been consolidated into a single event type. This
542 event will always contain a Module field specifying the name of the module
543 and a Status field denoting the result of the reload. All modules now issue
544 this event when being reloaded.
546 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
547 fail to receive this event due to being connected after modules have loaded.
548 AMI connections that want to know when Asterisk is ready should listen for
549 the 'FullyBooted' event.
551 * app_fax now sends the same send fax/receive fax events as res_fax. The
552 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
553 now the 'ReceiveFAX' event.
555 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
556 'MusicOnHoldStop'. The sub type field has been removed.
558 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
559 carrier for another protocol.
561 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
562 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
563 to the specific channel. 'Both' may be specified to play a tone to both
564 channels. The old 'yes' option is still accepted as a way of playing the
565 tone to Channel2 only.
567 * The AMI 'Status' response event to the AMI Status action replaces the
568 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
569 indicate what bridge the channel is currently in.
571 * The AMI 'Hold' event has been moved out of individual channel drivers, into
572 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
575 * The AMI events in app_queue have been made more consistent with each other.
576 Events that reference channels (QueueCaller* and Agent*) will show
577 information about each channel. The (infamous) 'Join' and 'Leave' AMI
578 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
580 * The 'MCID' AMI event now publishes a channel snapshot when available and
581 its non-channel-snapshot parameters now use either the "MCallerID" or
582 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
583 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
584 parameters in the channel snapshot.
586 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
587 'AgentLogin' and 'AgentLogoff' respectively.
589 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
590 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
592 * 'ChannelUpdate' events have been removed.
594 * All AMI events now contain a 'SystemName' field, if available.
596 * Local channel optimization is now conveyed in two events:
597 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
598 when the Local channel driver begins attempting to optimize itself out of
599 the media path; the End event is sent after the channel halves have
600 successfully optimized themselves out of the media path.
602 * Local channel information in events is now prefixed with 'LocalOne' and
603 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
604 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
605 and 'LocalOptimizationEnd' events.
607 * The option 'allowmultiplelogin' can now be set or overriden in a particular
608 account. When set in the general context, it will act as the default
609 setting for defined accounts.
611 * The 'BridgeAction' event was removed. It technically added no value, as the
612 Bridge Action already receives confirmation of the bridge through a
613 successful completion Event.
615 * The 'BridgeExec' events were removed. These events duplicated the events that
616 occur in the Briding API, and are conveyed now through BridgeCreate,
617 BridgeEnter, and BridgeLeave events.
619 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
620 previous versions. They now report all SR/RR packets sent/received, and
621 have been restructured to better reflect the data sent in a SR/RR. In
622 particular, the event structure now supports multiple report blocks.
624 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
625 raised when a blind transfer/attended transfer completes successfully.
626 They contain information about the transfer that just completed, including
627 the location of the transfered channel.
629 * Added a 'security' class to AMI which outputs the required fields for
630 security messages similar to the log messages from res_security_log
632 CDR (Call Detail Records)
634 * Significant changes have been made to the behavior of CDRs. The CDR engine
635 was effectively rewritten and built on the Stasis message bus. For a full
636 definition of CDR behavior in Asterisk 12, please read the specification
637 on the Asterisk wiki (wiki.asterisk.org).
639 * CDRs will now be created between all participants in a bridge. For each
640 pair of channels in a bridge, a CDR is created to represent the path of
641 communication between those two endpoints. This lets an end user choose who
642 to bill for what during bridge operations with multiple parties.
644 * The duration, billsec, start, answer, and end times now reflect the times
645 associated with the current CDR for the channel, as opposed to a cumulative
646 measurement of all CDRs for that channel.
648 * When a CDR is dispatched, user defined CDR variables from both parties are
649 included in the resulting CDR. If both parties have the same variable, only
650 the Party A value is provided.
652 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
653 information regarding the CDR engine is logged as verbose messages. This
654 option should only be used if the behavior of the CDR engine needs to be
657 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
658 normally configured in cdr.conf.
660 * Added CLI command 'cdr show active {channel}'. When {channel} is not
661 specified, this command provides a summary of the channels with CDR
662 information and their statistics. When {channel} is specified, it shows
663 detailed information about all records associated with {channel}.
665 CEL (Channel Event Logging)
667 * CEL has undergone significant rework in Asterisk 12, and is now built on the
668 Stasis message bus. Please see the specification for CEL on the Asterisk
669 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
672 * The 'extra' field of all CEL events that use it now consists of a JSON blob
673 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
675 * BLINDTRANSFER events now report the transferee bridge unique
676 identifier, extension, and context in a JSON blob as the extra string
677 instead of the transferee channel name as the peer.
679 * ATTENDEDTRANSFER events now report the peer as NULL and additional
680 information in the 'extra' string as a JSON blob. For transfers that occur
681 between two bridged channels, the 'extra' JSON blob contains the primary
682 bridge unique identifier, the secondary channel name, and the secondary
683 bridge unique identifier. For transfers that occur between a bridged channel
684 and a channel running an app, the 'extra' JSON blob contains the primary
685 bridge unique identifier, the secondary channel name, and the app name.
687 * LOCAL_OPTIMIZE events have been added to convey local channel
688 optimizations with the record occurring for the semi-one channel and
689 the semi-two channel name in the peer field.
691 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
692 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
693 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
694 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
695 regardless of whether or not that bridge happens to contain multiple
700 * When compiled with '--enable-dev-mode', the astobj2 library will now add
701 several CLI commands that allow for inspection of ao2 containers that
702 register themselves with astobj2. The CLI commands are 'astobj2 container
703 dump', 'astobj2 container stats', and 'astobj2 container check'.
705 * Added specific CLI commands for bridge inspection. This includes 'bridge
706 show all', which lists all bridges in the system, and 'bridge show {id}',
707 which provides specific information about a bridge.
709 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
710 ejecting the channels currently in the bridge. If the channels cannot
711 continue in the dialplan or application that put them in the bridge, they
714 * Added command 'bridge kick'. This will eject a single channel from a bridge.
716 * Added commands to inspect and manipulate the registered bridge technologies.
717 This include 'bridge technology show', which lists the registered bridge
718 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
719 which controls whether or not a registered bridge technology can be used
720 during smart bridge operations. If a technology is suspended, it will not
721 be used when a bridge technology is picked for channels; when unsuspended,
722 it can be used again.
724 * The command 'config show help {module} {type} {option}' will show
725 configuration documentation for modules with XML configuration
726 documentation. When {module}, {type}, and {option} are omitted, a listing
727 of all modules with registered documentation is displayed. When {module}
728 is specified, a listing of all configuration types for that module is
729 displayed, along with their synopsis. When {module} and {type} are
730 specified, a listing of all configuration options for that type are
731 displayed along with their synopsis. When {module}, {type}, and {option}
732 are specified, detailed information for that configuration option is
735 * Added 'core show sounds' and 'core show sound' CLI commands. These display
736 a listing of all installed media sounds available on the system and
737 detailed information about a sound, respectively.
739 * 'xmldoc dump' has been added. This CLI command will dump the XML
740 documentation DOM as a string to the specified file. The Asterisk core
741 will populate certain XML elements pulled from the source files with
742 additional run-time information; this command lets a user produce the
743 XML documentation with all information.
747 * Parking has been pulled from core and placed into a separate module called
748 res_parking. See Parking changes below for more details. Configuration for
749 parking should now be performed in res_parking.conf. Configuration for
750 parking in features.conf is now unsupported.
752 * Core attended transfers now have several new options. While performing an
753 attended transfer, the transferer now has the following options:
754 - *1 - cancel the attended transfer (configurable via atxferabort)
755 - *2 - complete the attended transfer, dropping out of the call
756 (configurable via atxfercomplete)
757 - *3 - complete the attended transfer, but stay in the call. This will turn
758 the call into a multi-party bridge (configurable via atxferthreeway)
759 - *4 - swap to the other party. Once an attended transfer has begun, this
760 options may be used multiple times (configurable via atxferswap)
762 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
763 must be on the channel initiating the transfer to have any effect.
765 * The BRIDGE_FEATURES channel variable would previously only set features for
766 the calling party and would set this feature regardless of whether the
767 feature was in caps or in lowercase. Use of a caps feature for a letter
768 will now apply the feature to the calling party while use of a lowercase
769 letter will apply that feature to the called party.
771 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
773 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
774 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
775 activated the dynamic feature.
777 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
778 only on the channel executing the dynamic feature. Executing a dynamic
779 feature on the bridge peer in a multi-party bridge will execute it on all
780 peers of the activating channel.
782 * You can now have the settings for a channel updated using the FEATURE()
783 and FEATUREMAP() functions inherited to child channels by setting
784 FEATURE(inherit)=yes.
786 * automixmon now supports additional channel variables from automon including:
787 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
788 and TOUCH_MIXMONITOR_MESSAGE_STOP
790 * A new general features.conf option 'recordingfailsound' has been added which
791 allowssetting a failure sound for a user tries to invoke a recording feature
792 such as automon or automixmon and it fails.
794 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
795 features.c for atxferdropcall=no to work properly. This option now just
800 * Added log rotation strategy 'none'. If set, no log rotation strategy will
801 be used. Given that this can cause the Asterisk log files to grow quickly,
802 this option should only be used if an external mechanism for log management
807 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
808 will store the path information for that peer when it registers. Realtime
809 tables can also use the 'supportpath' field to enable Path header support.
811 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
812 objectIdentifier. This maps to the supportpath option in sip.conf.
816 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
817 provides modules a useful abstraction on top of the many storage mechanisms
818 in Asterisk, including the Asterisk Database, static configuration files,
819 static Realtime, and dynamic Realtime. It also provides a caching service.
820 Users can configure a hierarchy of data storage layers for specific modules
823 * All future modules which utilize Sorcery for object persistence must have a
824 column named "id" within their schema when using the Sorcery realtime module.
825 This column must be able to contain a string of up to 128 characters in length.
827 Security Events Framework
829 * Security Event timestamps now use ISO 8601 formatted date/time instead of
830 the "seconds-microseconds" format that it was using previously.
834 * The Stasis message bus is a publish/subscribe message bus internal to
835 Asterisk. Many services in Asterisk are built on the Stasis message bus,
836 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
837 Stasis can be configured in stasis.conf. Note that these parameters operate
838 at a very low level in Asterisk, and generally will not require changes.
842 * When a channel driver is configured to enable jiterbuffers, they are now
843 applied unconditionally when a channel joins a bridge. If a jitterbuffer
844 is already set for that channel when it enters, such as by the JITTERBUFFER
845 function, then the existing jitterbuffer will be used and the one set by
846 the channel driver will not be applied.
850 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
851 dialplan applications provided by the app_agent_pool module. Agents are
852 connected with callers using the new AgentRequest dialplan application.
853 The Agents:<agent-id> device state is available to monitor the status of an
854 agent. See agents.conf.sample for valid configuration options.
856 * The updatecdr option has been removed. Altering the names of channels on a
857 CDR is not supported - the name of the channel is the name of the channel,
858 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
859 has also been removed, for the same reason.
861 * The endcall and enddtmf configuration options are removed. Use the
862 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
863 channel before calling AgentLogin.
867 * chan_bridge has been removed. Its functionality has been incorporated
868 directly into the ConfBridge application itself.
872 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
873 of the specified span and its B-channels. Note that this command should
874 only be used if you understand the risks it entails.
876 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
877 A range of channels can be specified to be destroyed. Note that this command
878 should only be used if you understand the risks it entails.
880 * Added the CLI command 'dahdi create channels'. A range of channels can be
881 specified to be created, or the keyword 'new' can be used to add channels
886 * IPv6 support has been added. We are now able to bind to and
887 communicate using IPv6 addresses.
891 * The /b option has been removed.
893 * chan_local moved into the system core and is no longer a loadable module.
897 * Added general support for busy detection.
899 * Added ECAM command support for Sony Ericsson phones.
903 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
904 SIP stack. A collection of resource modules provides the bulk of the SIP
905 functionality. For more information on the new SIP channel driver, see
906 https://wiki.asterisk.org/wiki/x/JYGLAQ
910 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
911 using the 'supportpath' setting, either on a global basis or on a peer basis.
912 This setting enables Asterisk to route outgoing out-of-dialog requests via a
913 set of proxies by using a pre-loaded route-set defined by the Path headers in
914 the REGISTER request. See Realtime updates for more configuration information.
916 * The SIP_CODEC family of variables may now specify more than one codec. Each
917 codec must be separated by a comma. The first codec specified is the
918 preferred codec for the offer. This allows a dialplan writer to specify both
919 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
921 * The 'callevents' parameter has been removed. Hold AMI events are now raised
922 in the core, and can be filtered out using the 'eventfilter' parameter
925 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
926 codecs configured for a peer instead of the requested codec.
928 * The option "register_retry_403" has been added to chan_sip to work around
929 servers that are known to erroneously send 403 in response to valid
930 REGISTER requests and allows Asterisk to continue attepmting to connect.
934 * Added the 'immeddialkey' parameter. If set, when the user presses the
935 configured key the already entered number will be immediately dialed. This
936 is useful when the dialplan allows for variable length pattern matching.
937 Valid options are '*' and '#'.
939 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
940 milliseconds) before a call forward is considered to not be answered.
942 * The 'serviceurl' parameter allows Service URLs to be attached to line
951 * The password option has been disabled, as the AgentLogin application no
952 longer provides authentication.
956 * Due to changes in the Asterisk core, this function is no longer needed to
957 preserve a MixMonitor on a channel during transfer operations and dialplan
958 execution. It is effectively obsolete.
962 * The 'amaflags' and 'accountcode' attributes for the CDR function are
963 deprecated. Use the CHANNEL function instead to access these attributes.
965 * The 'l' option has been removed. When reading a CDR attribute, the most
966 recent record is always used. When writing a CDR attribute, all non-finalized
969 * The 'r' option has been removed, for the same reason as the 'l' option.
971 * The 's' option has been removed, as LOCKED semantics no longer exist in the
976 * A new function CDR_PROP has been added. This function lets you set properties
977 on a channel's active CDRs. This function is write-only. Properties accept
978 boolean values to set/clear them on the channel's CDRs. Valid properties
980 - 'party_a' - make this channel the preferred Party A in any CDR between two
981 channels. If two channels have this property set, the creation time of the
982 channel is used to determine who is Party A. Note that dialed channels are
983 never Party A in a CDR.
984 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
985 application when set to True, and analogous to the 'e' option in ResetCDR
990 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
991 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
992 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
995 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
996 string, i.e., [[context],extension],priority. If set on a channel, if a
997 channel leaves a bridge but is not hung up it will resume dialplan execution
1002 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1003 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1004 The value of this setting is ignored when disabled is used for the argument.
1008 * A new function provided by chan_pjsip, this function can be used in
1009 conjunction with the Dial application to construct a dial string that will
1010 dial all contacts on an Address of Record associated with a chan_pjsip
1015 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1016 outbound channel prior to dialing.
1020 * Redirecting reasons can now be set to arbitrary strings. This means
1021 that the REDIRECTING dialplan function can be used to set the redirecting
1022 reason to any string. It also allows for custom strings to be read as the
1023 redirecting reason from SIP Diversion headers.
1027 * The SPEECH_ENGINE function now supports read operations. When read from, it
1028 will return the current value of the requested attribute.
1034 res_agi (Asterisk Gateway Interface)
1036 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1038 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1041 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1042 will start the playback of the audio at the position specified. It will
1043 also return the final position of the file in 'endpos'.
1045 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1046 channel variable if the user stopped the file playback or if a remote
1047 entity stopped the playback. If neither stopped the playback, it will
1048 indicate the overall success/failure of the playback. If stopped early,
1049 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1052 * The SAY ALPHA command now accepts an additional parameter to control
1053 whether it specifies the case of uppercase, lowercase, or all letters to
1054 provide functionality similar to SayAlphaCase.
1056 res_ari (Asterisk RESTful Interface) (and others)
1058 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1059 control telephony primitives in Asterisk by remote client. This includes
1060 channels, bridges, endpoints, media, and other fundamental concepts. Users
1061 of ARI can develop their own communications applications, controlling
1062 multiple channels using an HTTP RESTful interface and receiving JSON events
1063 about the objects via a WebSocket connection. ARI can be configured in
1064 Asterisk via ari.conf. For more information on ARI, see
1065 https://wiki.asterisk.org/wiki/x/0YCLAQ
1069 * Parking has been extracted from the Asterisk core as a loadable module,
1070 res_parking. Configuration for parking is now provided by res_parking.conf.
1071 Configuration through features.conf is no longer supported.
1073 * res_parking uses the configuration framework. If an invalid configuration is
1074 supplied, res_parking will fail to load or fail to reload. Previously,
1075 invalid configurations would generally be accepted, with certain errors
1076 resulting in individually disabled parking lots.
1078 * Parked calls are now placed in bridges. While this is largely an
1079 architectural change, it does have implications on how channels in a parking
1080 lot are viewed. For example, commands that display channels in bridges will
1081 now also display the channels in a parking lot.
1083 * The order of arguments for the new parking applications have been modified.
1084 Timeout and return context/exten/priority are now implemented as options,
1085 while the name of the parking lot is now the first parameter. See the
1086 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1087 in-depth information as well as syntax.
1089 * Extensions are by default no longer automatically created in the dialplan to
1090 park calls or pickup parked calls. Generation of dialplan extensions can be
1091 enabled using the 'parkext' configuration option.
1093 * ADSI functionality for parking is no longer supported. The 'adsipark'
1094 configuration option has been removed as a result.
1096 * The PARKINGSLOT channel variable has been deprecated in favor of
1097 PARKING_SPACE to match the naming scheme of the new system.
1099 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1100 channel even when the configuration option 'comebactoorigin' is enabled.
1102 * A new CLI command 'parking show' has been added. This allows a user to
1103 inspect the parking lots that are currently in use.
1104 'parking show <parkinglot>' will also show the parked calls in a specific
1107 * The CLI command 'parkedcalls' is now deprecated in favor of
1108 'parking show <parkinglot>'.
1110 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1111 can be used to get a list of parked calls for a specific parking lot.
1113 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1114 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1115 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1116 longer a required argument.
1118 * The ParkAndAnnounce application is now provided through res_parking instead
1119 of through the separate app_parkandannounce module.
1121 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1122 by default. Instead, it will follow the timeout rules of the parking lot. The
1123 old behavior can be reproduced by using the 'c' option.
1125 * Dynamic parking lots will now fail to be created under the following
1127 - if the parking lot specified by PARKINGDYNAMIC does not exist
1128 - if they require exclusive park and parkedcall extensions which overlap
1129 with existing parking lots.
1131 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1132 currently contain no calls. Dynamic parking lots containing parked calls
1133 will persist through the reloads without alteration.
1135 * If 'parkext_exclusive' is set for a parking lot and that extension is
1136 already in use when that parking lot tries to register it, this is now
1137 considered a parking system configuration error. Configurations which do
1138 this will be rejected.
1140 * Added channel variable PARKER_FLAT. This contains the name of the extension
1141 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1142 comebacktoorigin is disabled, but the dialplan or an external control
1143 mechanism wants to use the extension in the park-dial context that was
1144 generated to re-dial the parker on timeout.
1146 res_pjsip (and many others)
1148 * A large number of resource modules make up the SIP stack based on pjsip.
1149 The chan_pjsip channel driver users these resource modules to provide
1150 various SIP functionality in Asterisk. The majority of configuration for
1151 these modules is performed in pjsip.conf. Other modules may use their
1152 own configuration files.
1156 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1157 them, an Asterisk-specific version of PJSIP needs to be installed.
1158 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1160 res_statsd/res_chan_stats
1162 * A new resource module, res_statsd, has been added, which acts as a statsd
1163 client. This module allows Asterisk to publish statistics to a statsd
1164 server. In conjunction with res_chan_stats, it will publish statistics about
1165 channels to the statsd server. It can be configured via res_statsd.conf.
1169 * Device state for XMPP buddies is now available using the following format:
1170 XMPP/<client name>/<buddy address>
1171 If any resource is available the device state is considered to be not in use.
1172 If no resources exist or all are unavailable the device state is considered
1179 Realtime/Database Scripts
1181 * Asterisk previously included example db schemas in the contrib/realtime/
1182 directory of the source tree. This has been replaced by a set of database
1183 migrations using the Alembic framework. This allows you to use alembic to
1184 initialize the database for you. It will also serve as a database migration
1185 tool when upgrading Asterisk in the future.
1187 See contrib/ast-db-manage/README.md for more details.
1191 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1192 This python script will convert an existing sip.conf file to a
1193 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1194 is meant to be an aid in converting an existing chan_sip configuration to
1195 a chan_pjsip configuration, but it is expected that configuration beyond
1196 what the script provides will be needed.
1199 ------------------------------------------------------------------------------
1200 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1201 ------------------------------------------------------------------------------
1205 * The Asterisk build system will now build and install a shared library
1206 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1207 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1208 that Asterisk can ensure that these functions do *not* get called by any
1209 modules that are loaded into Asterisk, since they should only be called once
1210 in any single process. If desired, this feature can be disabled by supplying
1211 the "--disable-asteriskssl" option to the configure script.
1213 * A new make target, 'full', has been added to the Makefile. This performs
1214 the same compilation actions as make all, but will also scan the entirety of
1215 each source file for documentation. This option is needed to generate AMI
1216 event documentation. Note that your system must have Python in order for
1217 this make target to succeed.
1219 * The optimization portion of the build system has been reworked to avoid
1220 broken builds on certain architectures. All architecture-specific
1221 optimization has been removed in favor of using -march=native to allow gcc
1222 to detect the environment in which it is running when possible. This can
1223 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1225 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1226 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1228 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1229 previously parsed the header file to obtain the version of Asterisk, you
1230 will now have to go through Asterisk to get the version information.
1238 * Added 'F()' option. Similar to the dial option, this can be supplied with
1239 arguments indicating where the callee should go after the caller is hung up,
1240 or without options specified, the priority after the Queue will be used.
1245 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1246 all non-admin participants on a conference. The confbridge configuration
1247 file also allows for the default sounds played to all conference users when
1248 this occurs to be overriden using sound_participants_unmuted and
1249 sound_participants_muted.
1251 * Added menu action participant_count. This will playback the number of
1252 current participants in a conference.
1254 * Added announcement configuration option to user profile. If set the sound
1255 file will be played to the user, and only the user, upon joining the
1258 * Added record_file_append option that defaults to "yes", but if set to no
1259 will create a new file between each start/stop recording.
1264 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1265 channels respectively before the callee channels are called.
1270 * Added support for IPv6.
1272 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1273 external process will cause the current playlist to be cleared, including
1274 stopping any audio file that is currently playing. This is useful when you
1275 want to interrupt audio playback only when specific DTMF is entered by the
1281 * A new option, 'I' has been added to app_followme. By setting this option,
1282 Asterisk will not update the caller with connected line changes when they
1283 occur. This is similar to app_dial and app_queue.
1285 * The 'N' option is now ignored if the call is already answered.
1287 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1288 and caller channels respectively before the callee channels are called.
1290 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1296 * MixMonitor hooks now have IDs associated with them which can be used to
1297 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1298 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1299 now accepts that ID as an argument.
1301 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1302 indicated mailboxes.
1307 * The connect action in app_mysql now allows you to specify a port number to
1308 connect to. This is useful if you run a MySQL server on a non-standard
1314 * Increased the default number of allowed destinations from 5 to 12.
1319 * The app_page application now no longer depends on DAHDI or app_meetme. It
1320 has been re-architected to use app_confbridge internally.
1325 * Added queue options autopausebusy and autopauseunavail for automatically
1326 pausing a queue member when their device reports busy or congestion.
1328 * The 'ignorebusy' option for queue members has been deprecated in favor of
1329 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1330 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1331 per interface basis. Individual ringinuse values can now be set in
1332 queues.conf via an argument to member definitions. Lastly, the queue
1333 'ringinuse' setting now only determines defaults for the per member
1334 'ringinuse' setting and does not override per member settings like it does
1335 in earlier versions.
1337 * Added 'F()' option. Similar to the dial option, this can be supplied with
1338 arguments indicating where the callee should go after the caller is hung up,
1339 or without options specified, the priority after the Queue will be used.
1341 * Added new option log_member_name_as_agent, which will cause the membername to
1342 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1343 state_interface has been set.
1345 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1347 * App_queue will now play periodic announcements for the caller that
1348 holds the first position in the queue while waiting for answer.
1352 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1353 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1354 changed arguments to SayUnixTime so that every option is truly optional even
1355 when using multiple options (so that j option could be used without having to
1356 manually specify timezone and format) There are other benefits, e.g., format
1357 can now be used without specifying time zone as well.
1362 * Addition of the VM_INFO function - see Function changes.
1364 * The imapserver, imapport, and imapflags configuration options can now be
1365 overriden on a user by user basis.
1367 * When voicemail plays a message's envelope with saycid set to yes, when
1368 reaching the caller id field it will play a recording of a file with the same
1369 base name as the sender's callerid if there is a similarly named file in
1370 <astspooldir>/recordings/callerids/
1372 * Voicemails now contains a unique message identifier "msg_id", which is stored
1373 in the message envelope with the sound files. IMAP backends will now store
1374 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1375 backends will store the message identifier in a "msg_id" column. See
1376 UPGRADE.txt for more information.
1378 * Added VoiceMailPlayMsg application. This application will play a single
1379 voicemail message from a mailbox. The result of the application, SUCCESS or
1380 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1385 * Hangup handlers can be attached to channels using the CHANNEL() function.
1386 Hangup handlers will run when the channel is hung up similar to the h
1387 extension. The hangup_handler_push option will push a GoSub compatible
1388 location in the dialplan onto the channel's hangup handler stack. The
1389 hangup_handler_pop option will remove the last added location, and optionally
1390 replace it with a new GoSub compatible location. The hangup_handler_wipe
1391 option will remove all locations on the stack, and optionally add a new
1394 * The expression parser now recognizes the ABS() absolute value function,
1395 which will convert negative floating point values to positive values.
1397 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1398 control of faxdetect.
1400 * Addition of the VM_INFO function that can be used to retrieve voicemail
1401 user information, such as the email address and full name.
1402 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1405 * The REDIRECTING function now supports the redirecting original party id
1408 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1409 lets you set some of the configuration options from the [general] section
1410 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1411 the key sequence used to activate built-in features, such as blindxfer,
1412 and automon. See the built-in documentation for details.
1414 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1415 instead of simply the uri. This is the format that MessageSend() can use
1416 in the from parameter for outgoing SIP messages.
1418 * Added the PRESENCE_STATE function. This allows retrieving presence state
1419 information from any presence state provider. It also allows setting
1420 presence state information from a CustomPresence presence state provider.
1421 See AMI/CLI changes for related commands.
1423 * Added the AMI_CLIENT function to make manager account attributes available
1424 to the dialplan. It currently supports returning the current number of
1425 active sessions for a given account.
1427 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1428 and the REDIRECTING functions.
1436 * Added a manager event "LocalBridge" for local channel call bridges between
1437 the two pseudo-channels created.
1442 * Added dialtone_detect option for analog ports to disconnect incoming
1443 calls when dialtone is detected.
1445 * Added option colp_send to send ISDN connected line information. Allowed
1446 settings are block, to not send any connected line information; connect, to
1447 send connected line information on initial connect; and update, to send
1448 information on any update during a call. Default is update.
1450 * Add options namedcallgroup and namedpickupgroup to support installations
1451 where a higher number of groups (>64) is required.
1453 * Added support to use private party ID information with PRI calls.
1458 * A new channel driver named chan_motif has been added which provides support for
1459 Google Talk and Jingle in a single channel driver. This new channel driver includes
1460 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1461 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1462 specification, current Google Jingle specification, and the original Google Talk
1468 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1469 globally and overriden on a peer by peer basis.
1471 * Direct media functionality has been added. Options in config are:
1472 directmedia (directrtp) and directrtpsetup (earlydirect)
1474 * ChannelUpdate events now contain a CallRef header.
1479 * Asterisk will no longer substitute CID number for CID name in the display
1480 name field if CID number exists without a CID name. This change improves
1481 compatibility with certain device features such as Avaya IP500's directory
1484 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1485 created using that setting to not be removed during SIP reload.
1487 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1488 request with a "Record:" header, this will turn the requested feature on/off.
1489 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1490 dynamic features must be enabled and configured properly on the requesting
1491 channel for this to function properly.
1493 * Add support to realtime for the 'callbackextension' option.
1495 * When multiple peers exist with the same address, but differing
1496 callbackextension options, incoming requests that are matched by address
1497 will be matched to the peer with the matching callbackextension if it is
1500 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1501 which set the force_rport and comedia options automatically if Asterisk
1502 detects that an incoming SIP request crossed a NAT after being sent by
1503 the remote endpoint.
1505 * The default global nat setting in sip.conf has been changed from force_rport
1506 to auto_force_rport.
1508 * NAT settings are now a combinable list of options. The equivalent of the
1509 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1511 * Adds an option send_diversion which can be disabled to prevent
1512 diversion headers from automatically being added to INVITE requests.
1514 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1515 be sent to the remote host at a given interval to keep the NAT mapping open.
1516 This can be enabled using the keepalive configuration option.
1518 * Add option 'tonezone' to specify country code for indications. This option
1519 can be set both globally and overridden for specific peers.
1521 * The SIP Security Events Framework now supports IPv6.
1523 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1524 between multiple user agents. When set, for directmedia reinvites,
1525 Asterisk will not send an immediate reinvite on an incoming call leg. This
1526 option is useful when peered with another SIP user agent that is known to
1527 send immediate direct media reinvites upon call establishment.
1529 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1532 * Add options subminexpiry and submaxexpiry to set limits of subscription
1533 timer independently from registration timer settings. The setting of the
1534 registration timer limits still is done by options minexpiry, maxexpiry
1535 and defaultexpiry. For backwards compatibility the setting of minexpiry
1536 and maxexpiry also is used to configure the subscription timer limits if
1537 subminexpiry and submaxexpiry are not set in sip.conf.
1539 * Set registration timer limits to default values when reloading sip
1540 configuration and values are not set by configuration.
1542 * Add options namedcallgroup and namedpickupgroup to support installations
1543 where a higher number of groups (>64) is required.
1545 * When a MESSAGE request is received, the address the request was received from
1546 is now saved in the SIP_RECVADDR variable.
1548 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1549 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1550 the ANI2/OLI information is set on the channel, which can be retrieved using
1551 the CALLERID function.
1553 * Peers can now be configured to support negotiation of ICE candidates using
1554 the setting icesupport. See res_rtp_asterisk changes for more information.
1556 * Added support for format attribute negotiation. See the Codecs changes for
1559 * Extra headers specified with SIPAddHeader are sent with the REFER message
1560 when using Transfer application. See refer_addheaders in sip.conf.sample.
1562 * Added support to use private party ID information with calls.
1564 * Adds an option discard_remote_hold_retrieval that when set stops telling
1565 the peer to start music on hold.
1570 * Added skinny version 17 protocol support.
1574 --------------------
1575 * Added ability to use multiple lines for a single phone. This allows multiple
1576 calls to occur on a single phone, using callwaiting and switching between calls.
1578 * Added option 'sharpdial' allowing end dialing by pressing # key
1580 * Added option 'interdigit_timer' to control phone dial timeout
1582 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1584 * Added global 'debug' option, that enables debug in channel driver
1586 * Added ability to translate on-screen menu in multiple languages. Tested on
1587 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1588 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1591 * In addition to English added French and Russian languages for on-screen menus
1593 * Reworked dialing number input: added dialing by timeout, immediate dial on
1594 on dialplan compare, phone number length now not limited by screen size
1596 * Added ability to pickup a call using features.conf defined value and
1602 * Add options namedcallgroup and namedpickupgroup to support installations
1603 where a higher number of groups (>64) is required.
1605 * Added support to use private party ID information with calls.
1610 * The minimum DTMF duration can now be configured in asterisk.conf
1611 as "mindtmfduration". The default value is (as before) set to 80 ms.
1612 (previously it was only available in source code)
1614 * Named ACLs can now be specified in acl.conf and used in configurations that
1615 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1616 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1617 working ACL. In addition, some CLI commands have been added to provide
1618 show information and allow for module reloading - see CLI Changes.
1620 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1621 items (separated by commas), and items in the rule can be negated by prefixing
1622 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1623 longer necessray to control the order that the 'permit' and 'deny' columns are
1624 returned from queries.
1626 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1627 be used within the dynamic weight attribute when specifying a mapping.
1629 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1630 header, instead of putting the user defined event name there. When enabled
1631 the UserDefType header is added for user defined events. This feature is
1632 enabled with the setting show_user_defined.
1634 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1635 line purposes use the following variables instead of their macro equivalents:
1636 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1637 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1638 cc_callback_macro in channel configurations.
1640 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1643 * Call files now support the "early_media" option to connect with an outgoing
1644 extension when early media is received.
1646 * Added support to use private party ID information with calls.
1651 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1652 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1653 AGI application would exit immediately after a channel hangup is detected.
1655 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1656 are resolved and each address is attempted in turn until one succeeds or
1660 AMI (Asterisk Manager Interface)
1662 * The originate action now has an option "EarlyMedia" that enables the
1663 call to bridge when we get early media in the call. Previously,
1664 early media was disregarded always when originating calls using AMI.
1666 * Added setvar= option to manager accounts (much like sip.conf)
1668 * Originate now generates an error response if the extension given is not found
1671 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1672 them if the i(variable) option is used. StopMixMonitor will accept
1673 MixMonitorID as an option to close specific MixMonitors.
1675 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1676 updated to include information about peers configured with
1677 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1678 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1679 returned if auto_force_rport is not enabled.
1681 * Added SIPpeerstatus manager command which will generate PeerStatus events
1682 similar to the existing PeerStatus events found in chan_sip on demand.
1684 * Hangup now can take a regular expression as the Channel option. If you want
1685 to hangup multiple channels, use /regex/ as the Channel option. Existing
1686 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1687 the manager will send you a list of channels back that were hung up.
1689 * Support for IPv6 addresses has been added.
1691 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1692 documentation is only generated when Asterisk is compiled using 'make full'.
1693 See the CLI section for commands to display AMI event information.
1695 * The AMI Hangup event now includes the AccountCode header so you can easily
1696 correlate with AMI Newchannel events.
1698 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1699 the StateInterface of the queue member.
1701 * Added AMI event SessionTimeout in the Call category that is issued when a
1702 call is terminated due to either RTP stream inactivity or SIP session timer
1705 * CEL events can now contain a user defined header UserDefType. See core
1706 changes for more information.
1708 * OOH323 ChannelUpdate events now contain a CallRef header.
1710 * Added PresenceState command. This command will report the presence state for
1711 the given presence provider.
1713 * Added Parkinglots command. This will list all parking lots as a series of
1714 AMI Parkinglot events.
1716 * Added MessageSend command. This behaves in the same manner as the
1717 MessageSend application, and is a technolgoy agnostic mechanism to send out
1718 of call text messages.
1720 * Added "message" class authorization. This grants an account permission to
1721 send out of call messages. Write-only.
1726 * The "dialplan add include" command has been modified to create context a context
1727 if one does not already exist. For instance, "dialplan add include foo into bar"
1728 will create context "bar" if it does not already exist.
1730 * A "dialplan remove context" command has been added to remove a context from
1733 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1734 filenames of all running mixmonitors on a channel.
1736 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1737 numeric instead of 0, 1, or 2.
1739 * "stun show status" will show a table describing how the STUN client is
1742 * "acl show [named acl]" will show information regarding a Named ACL. The
1743 acl module can be reloaded with "reload acl".
1745 * Added CLI command to display AMI event information - "manager show events",
1746 which shows a list of all known and documented AMI events, and "manager show
1747 event [event name]", which shows detail information about a specific AMI
1750 * The result of the CLI command "queue show" now includes the state interface
1751 information of the queue member.
1753 * The command "core set verbose" will now set a separate level of logging for
1754 each remote console without affecting any other console.
1756 * Added command "cdr show pgsql status" to check connection status
1758 * "sip show channel" will now display the complete route set.
1760 * Added "presencestate list" command. This command will list all custom
1761 presence states that have been set by using the PRESENCE_STATE dialplan
1764 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1765 command. This changes a custom presence to a new state.
1770 * Codec lists may now be modified by the '!' character, to allow succinct
1771 specification of a list of codecs allowed and disallowed, without the
1772 requirement to use two different keywords. For example, to specify all
1773 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1775 * Add support for parsing SDP attributes, generating SDP attributes, and
1776 passing it through. This support includes codecs such as H.263, H.264, SILK,
1777 and CELT. You are able to set up a call and have attribute information pass.
1778 This should help considerably with video calls.
1780 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1781 just like the GSM codec.
1785 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1786 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1790 * Asterisk version and build information is now logged at the beginning of a
1793 * Threads belonging to a particular call are now linked with callids which get
1794 added to any log messages produced by those threads. Log messages can now be
1795 easily identified as involved with a certain call by looking at their call id.
1796 Call ids may also be attached to log messages for just about any case where
1797 it can be determined to be related to a particular call.
1799 * Each logging destination and console now have an independent notion of the
1800 current verbosity level. Logger.conf now allows an optional argument to
1801 the 'verbose' specifier, indicating the level of verbosity sent to that
1802 particular logging destination. Additionally, remote consoles now each
1803 have their own verbosity level. The command 'core set verbose' will now set
1804 a separate level for each remote console without affecting any other
1810 * Added 'announcement' option which will play at the start of MOH and between
1811 songs in modes of MOH that can detect transitions between songs (eg.
1817 * New per parking lot options: comebackcontext and comebackdialtime. See
1818 configs/features.conf.sample for more details.
1820 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1823 * Channel variable PARKEDCALL is now set with the name of the parking lot
1824 when a timeout occurs.
1830 CDR Postgresql Driver
1832 * Added command "cdr show pgsql status" to check connection status
1835 CDR Adaptive ODBC Driver
1837 * Added schema option for databases that support specifying a schema.
1845 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1846 CALENDAR_WRITE has completed successfully.
1851 * A new option, 'probation' has been added to rtp.conf
1852 RTP in strictrtp mode can now require more than 1 packet to exit learning
1853 mode with a new source (and by default requires 4). The probation option
1854 allows the user to change the required number of packets in sequence to any
1855 desired value. Use a value of 1 to essentially restore the old behavior.
1856 Also, with strictrtp on, Asterisk will now drop all packets until learning
1857 mode has successfully exited. These changes are based on how pjmedia handles
1858 media sources and source changes.
1860 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1861 enabled or disabled using the icesupport setting. A variety of other
1862 settings have been introduced to configure STUN/TURN connections.
1867 * A new module, res_corosync, has been introduced. This module uses the
1868 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1869 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1870 Device State (presence) information. This module is very similar to, and
1871 is a replacement for the res_ais module that was in previous releases of
1877 * This module adds a cleaned up, drop-in replacement for res_jabber called
1878 res_xmpp. This provides the same externally facing functionality but is
1879 implemented differently internally. res_jabber has been deprecated in favor
1880 of res_xmpp; please see the UPGRADE.txt file for more information.
1885 * The safe_asterisk script has been updated to allow several of its parameters
1886 to be set from environment variables. This also enables a custom run
1887 directory of Asterisk to be specified, instead of defaulting to /tmp.
1889 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1890 its value to determine the directory to assume is the top-level directory of
1891 the source tree. If the variable is not set, it defaults to the current
1892 behavior and uses the current working directory.
1894 ------------------------------------------------------------------------------
1895 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1896 ------------------------------------------------------------------------------
1900 * Asterisk now has protocol independent support for processing text messages
1901 outside of a call. Messages are routed through the Asterisk dialplan.
1902 SIP MESSAGE and XMPP are currently supported. There are options in
1903 jabber.conf and sip.conf to allow enabling these features.
1904 -> jabber.conf: see the "sendtodialplan" and "context" options.
1905 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1906 and "outofcall_message_context" options.
1907 The MESSAGE() dialplan function and MessageSend() application have been
1908 added to go along with this functionality. More detailed usage information
1909 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1910 * If real-time text support (T.140) is negotiated, it will be preferred for
1911 sending text via the SendText application. For example, via SIP, messages
1912 that were once sent via the SIP MESSAGE request would be sent via RTP if
1913 T.140 text is negotiated for a call.
1917 * parkedmusicclass can now be set for non-default parking lots.
1919 Asterisk Manager Interface
1920 --------------------------
1921 * PeerStatus now includes Address and Port.
1922 * Added Hold events for when the remote party puts the call on and off hold
1923 for chan_dahdi ISDN channels.
1924 * Added new action MeetmeListRooms to list active conferences (shows same
1925 data as "meetme list" at the CLI).
1926 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1927 Description field that is set by 'description' in the channel configuration
1929 * Added Uniqueid header to UserEvent.
1930 * Added new action FilterAdd to control event filters for the current session.
1931 This requires the system permission and uses the same filter syntax as
1932 filters that can be defined in manager.conf
1933 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1934 versions had some instances of the event converted, but others were left
1935 as-is. All Unlink events should now be converted to Bridge events. The AMI
1936 protocol version number was incremented to 1.2 as a result of this change.
1938 Asterisk HTTP Server
1939 --------------------------
1940 * The HTTP Server can bind to IPv6 addresses.
1943 --------------------------
1944 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1945 with busydetect. usage example: busypattern=200,200,200,600
1948 --------------------------
1949 * New 'gtalk show settings' command showing the current settings loaded from
1951 * The 'logger reload' command now supports an optional argument, specifying an
1952 alternate configuration file to use.
1953 * 'dialplan add extension' command will now automatically create a context if
1954 the specified context does not exist with a message indicated it did so.
1955 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1956 Description field which can be populated with 'description' in the channel
1957 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1960 --------------------------
1961 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1962 thus allowing records which do NOT match the specified filter.
1963 * Added ability to log CONGESTION calls to CDR
1966 --------------------------
1967 * Ability to define custom SILK formats in codecs.conf.
1968 * Addition of speex32 audio format with translation.
1969 * CELT codec pass-through support and ability to define
1970 custom CELT formats in codecs.conf.
1971 * Ability to read raw signed linear files with sample rates
1972 ranging from 8khz - 192khz. The new file extensions introduced
1973 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1974 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1975 Skinny, H.323, etc) can still only support the following codecs:
1976 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1977 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1978 Video: h261, h263, h263p, h264, mpeg4
1983 --------------------------
1984 * New highly optimized and customizable ConfBridge application capable of
1985 mixing audio at sample rates ranging from 8khz-96khz.
1986 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1987 and bridge profiles on a channel.
1988 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1989 about a conference such as locked status and number of parties, admins,
1991 * Addition of video_mode option in confbridge.conf for adding video support
1992 into a bridge profile.
1993 * Addition of the follow_talker video_mode in confbridge.conf. This video
1994 mode dynamically switches the video feed to always display the loudest talker
1995 supplying video in the conference.
1999 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2000 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2001 variables from asterisk.conf.
2005 * Addition of the JITTERBUFFER dialplan function. This function allows
2006 for jitterbuffering to occur on the read side of a channel. By using
2007 this function conference applications such as ConfBridge and MeetMe can
2008 have the rx streams jitterbuffered before conference mixing occurs.
2009 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2011 * Added STRREPLACE function. This function let's the user search a variable
2012 for a given string to replace with another string as many times as the
2013 user specifies or just throughout the whole string.
2014 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2015 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2016 * Added extensions to chan_ooh323 in function CHANNEL()
2018 libpri channel driver (chan_dahdi) DAHDI changes
2019 --------------------------
2020 * Added moh_signaling option to specify what to do when the channel's bridged
2021 peer puts the ISDN channel on hold.
2022 * Added display_send and display_receive options to control how the display ie
2023 is handled. To send display text from the dialplan use the SendText()
2024 application when the option is enabled.
2025 * Added mcid_send option to allow sending a MCID request on a span.
2028 --------------------------
2029 * Added setvar option to calendar.conf to allow setting channel variables on
2030 notification channels.
2031 * Added "calendar show types" CLI command to list registered calendar
2035 --------------------------
2036 * Added two new options, r and t with file name arguments to record
2037 single direction (unmixed) audio recording separate from the bidirectional
2038 (mixed) recording. The mixed file name argument is optional now as long
2039 as at least one recording option is used.
2042 --------------------------
2043 * Added a new option, l, which will disable local call optimization for
2044 channels involved with the FollowMe thread. Use this option to improve
2045 compatability for a FollowMe call with certain dialplan apps, options, and
2049 --------------------------
2050 * Added option "k" that will automatically close the conference when there's
2051 only one person left when a user exits the conference.
2054 --------------------------
2055 * cel_pgsql now supports the 'extra' column for data added using the
2056 CELGenUserEvent() application.
2059 --------------------------
2060 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2061 in the sample extensions.lua file for syntax details.
2062 * Applications that perform jumps in the dialplan such as Goto will now
2063 execute properly. When pbx_lua detects that the context, extension, or
2064 priority we are executing on has changed it will immediately return control
2065 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2066 the priority after the currently executing priority.
2067 * An autoservice is now started by default for pbx_lua channels. It can be
2068 stopped and restarted using the autoservice_stop() and autoservice_start()
2072 --------------------------
2073 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2074 into a FAXStatus event with an 'Operation' header that will be either
2075 'send', 'receive', and 'gateway'.
2076 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2077 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2078 feature will handle converting a fax call between an audio T.30 fax terminal
2079 and an IFP T.38 fax terminal.
2083 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2084 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2085 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2089 * Added general option negative_penalty_invalid default off. when set
2090 members are seen as invalid/logged out when there penalty is negative.
2091 for realtime members when set remove from queue will set penalty to -1.
2092 * Added queue option autopausedelay when autopause is enabled it will be
2093 delayed for this number of seconds since last successful call if there
2094 was no prior call the agent will be autopaused immediately.
2095 * Added member option ignorebusy this when set and ringinuse is not
2096 will allow per member control of multiple calls as ringinuse does for
2101 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2103 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2104 one participant left (much like a normal call bridge)
2105 * Added extra argument to Originate to set timeout.
2109 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2110 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2111 utility in the UTILS section of menuselect. If an existing astdb is found and no
2112 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2113 convert an existing astdb to the SQLite3 version automatically at runtime.
2117 * Modules marked as deprecated are no longer marked as building by default. Enabling
2118 these modules is still available via menuselect.
2122 * authdebug is now disabled by default. To enable this functionaility again
2123 set authdebug = yes in iax.conf.
2127 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2128 releases it was disabled.
2132 * The PBX core previously made a call with a non-existing extension test for
2133 extension s@default and jump there if the extension existed.
2134 This was a bad default behaviour and violated the principle of least surprise.
2135 It has therefore been changed in this release. It may affect some
2136 applications and configurations that rely on this behaviour. Most channel
2137 drivers have avoided this for many releases by testing whether the extension
2138 called exists before starting the PBX and generating a local error.
2139 This behaviour still exists and works as before.
2141 Extension "s" is used when no extension is given in a channel driver,
2142 like immediate answer in DAHDI or calling to a domain with no user part
2145 ------------------------------------------------------------------------------
2146 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2147 ------------------------------------------------------------------------------
2151 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2152 now defaults to force_rport. It is very important that phones requiring nat=no be
2153 specifically set as such instead of relying on the default setting. If at all
2154 possible, all devices should have nat settings configured in the general section as
2155 opposed to configuring nat per-device.
2156 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2157 codecs sent in response to an INVITE to the single most preferred codec.
2158 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2159 to be used for the outgoing call. It must be one of the codecs configured
2161 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2162 to be used for holding a private key. If tlsprivatekey is not specified,
2163 tlscertfile is searched for both public and private key.
2164 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2165 outbound client connections to be specified.
2166 * The sendrpid parameter has been expanded to include the options
2167 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2168 header to be sent (equivalent to setting sendrpid=yes) and setting
2169 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2170 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2171 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2172 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2173 will accept the SDP even if the SDP version number is not properly incremented,
2174 but will generate a warning in the log indicating that the SIP peer that sent
2175 the SDP should have the 'ignoresdpversion' option set.
2176 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2177 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2178 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2179 remote side requests it and disables symmetric RTP support. Setting it to
2180 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2181 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2182 and enables symmetric RTP support.
2183 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2184 response. This permits the master channel to know how each channel dialled
2185 in a multi-channel setup resolved in an individual way. This carries a
2186 performance penalty and can be disabled in sip.conf using the
2187 'storesipcause' option.
2188 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2189 configuration for the externip and externhost options when tcp or tls is used.
2190 * Added support for message body (stored in content variable) to SIP NOTIFY message
2191 accessible via AMI and CLI.
2192 * Added 'media_address' configuration option which can be used to explicitly specify
2193 the IP address to use in the SDP for media (audio, video, and text) streams.
2194 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2195 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2197 * Added 'use_q850_reason' configuration option for generating and parsing
2198 if available Reason: Q.850;cause=<cause code> header. It is implemented
2199 in some gateways for better passing PRI/SS7 cause codes via SIP.
2200 * When dialing SIP peers, a new component may be added to the end of the dialstring
2201 to indicate that a specific remote IP address or host should be used when dialing
2202 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2203 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2204 ability to selectively force bridged channels to also be encrypted is also
2205 implemented. Branching in the dialplan can be done based on whether or not
2206 a channel has secure media and/or signaling.
2207 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2209 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2210 Charge messages to snom phones.
2211 * Added support for G.719 media streams.
2212 * Added support for 16khz signed linear media streams.
2213 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2214 RTP has been outfitted with the same abilities.
2215 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2216 available in device configurations as well as in the dial plan.
2217 * Addition of the 'subscribe_network_change' option for turning on and off
2218 res_stun_monitor module support in chan_sip.
2219 * Addition of the 'auth_options_requests' option for turning on and off
2220 authentication for OPTIONS requests in chan_sip.
2224 * Add #tryinclude statement for config files. This provides the same
2225 functionality as the #include statement however an asterisk module will
2226 still load if the filename does not exist. Using the #include statement
2227 Asterisk will not allow the module to load.
2231 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2232 on realtime updates.
2233 * Added the ability for chan_iax2 to inform the dialplan whether or not
2234 encryption is being used. This interoperates with the SIP SRTP implementation
2235 so that a secure SIP call can be bridged to a secure IAX call when the
2236 dialplan requires bridged channels to be "secure".
2237 * Addition of the 'subscribe_network_change' option for turning on and off
2238 res_stun_monitor module support in chan_iax.
2243 * Added ability to preset channel variables on indicated lines with the setvar
2244 configuration option. Also, clearvars=all resets the list of variables back
2246 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2247 See configs/res_pktccops.conf for more information.
2249 XMPP Google Talk/Jingle changes
2250 -------------------------------
2251 * Added the externip option to gtalk.conf.
2252 * Added the stunaddr option to gtalk.conf which allows for the automatic
2253 retrieval of the external ip from a stun server.
2257 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2258 match to a partial channel name.
2259 * Added .m3u support for Mp3Player application.
2260 * Added progress option to the app_dial D() option. When progress DTMF is
2261 present, those values are sent immediately upon receiving a PROGRESS message
2262 regardless if the call has been answered or not.
2263 * Added functionality to the app_dial F() option to continue with execution
2264 at the current location when no parameters are provided.
2265 * Added the 'a' option to app_dial to answer the calling channel before any
2266 announcements or macros are executed.
2267 * Modified app_dial to set answertime when the called channel answers even if
2268 the called channel hangs up during playback of an announcement.
2269 * Modified app_dial 'r' option to support an additional parameter to play an
2270 indication tone from indications.conf
2271 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2272 to cycle through the next available channel. By default this is still '*'.
2273 * Added x() option to app_chanspy. This option allows DTMF to be set to
2274 exit the application.
2275 * The Voicemail application has been improved to automatically ignore messages
2276 that only contain silence.
2277 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2278 associated mailbox(es) to be greetings-only.
2279 * The ChanSpy application now has the 'S' option, which makes the application
2280 automatically exit once it hits a point where no more channels are available
2282 * The ChanSpy application also now has the 'E' option, which spies on a single
2283 channel and exits when that channel hangs up.
2284 * The MeetMe application now turns on the DENOISE() function by default, for
2285 each participant. In our tests, this has significantly decreased background
2286 noise (especially noisy data centers).
2287 * Voicemail now permits storage of secrets in a separate file, located in the
2288 spool directory of each individual user. The control for this is located in
2289 the "passwordlocation" option in voicemail.conf. Please see the sample
2290 configuration for more information.
2291 * The ChanIsAvail application now exposes the returned cause code using a separate
2292 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2293 * Added 'd' option to app_followme. This option disables the "Please hold"
2295 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2296 received will terminate recording.
2297 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2298 Previously the folder could only be set per context, but has now been extended
2299 using the imapfolder option.
2300 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2301 * Voicemail now allows the pager date format to be specified separately from the
2303 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2304 to allow joining, leaving, and sending text to group chats.
2305 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2306 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2307 to all paged phones (and optionally excluding the caller's one using the new
2308 option 'n') before the call is bridged.
2309 * The 'f' option to Dial has been augmented to take an optional argument. If no
2310 argument is provided, the 'f' option works as it always has. If an argument is
2311 provided, then the connected party information of all outgoing channels created
2312 during the Dial will be set to the argument passed to the 'f' option.
2313 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2315 * The OSP lookup application adds in/outbound network ID, optional security,
2316 number portability, QoS reporting, destination IP port, custom info and service
2318 * Added new application VMSayName that will play the recorded name of the voicemail
2319 user if it exists, otherwise will play the mailbox number.
2320 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2321 retrieve state for a particular bridge, where <name> is the conference name
2322 * app_directory now allows exiting at any time using the operator or pound key.
2323 * Voicemail now supports setting a locale per-mailbox.
2324 * Two new applications are provided for declining counting phrases in multiple
2325 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2327 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2329 * Voicemail now includes rdnis within msgXXXX.txt file.
2330 * ExternalIVR now supports IPv6 addresses.
2331 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2332 at https://wiki.asterisk.org/wiki/x/oQBB
2333 * ParkedCall and Park can now specify the parking lot to use.
2337 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2338 over SRV records associated with a specific service. From the CLI, type
2339 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2340 details on how these may be used.
2341 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2342 pitch of a channel's tx and rx audio streams.
2343 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2344 setting various connected line and redirecting party information.
2345 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2346 support ISDN subaddressing.
2347 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2348 * For DAHDI channels, the CHANNEL() dialplan function now allows
2349 the dialplan to request changes in the configuration of the active
2350 echo canceller on the channel (if any), for the current call only.
2353 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2355 The possible values are:
2357 on - normal mode (the echo canceller is actually reinitialized)
2359 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2361 voice - voice mode (returns from FAX mode, reverting the changes that
2362 were made when FAX mode was requested)
2363 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2364 and setting variables on the channel which created the current channel.
2365 Administrators should take care to avoid naming conflicts, when multiple
2366 channels are dialled at once, especially when used with the Local channel
2367 construct (which all could set variables on the master channel). Usage
2368 of the HASH() dialplan function, with the key set to the name of the slave
2369 channel, is one approach that will avoid conflicts.
2370 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2372 * func_odbc now allows multiple row results to be retrieved without using
2373 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2374 from the same query by using the name of the function which retrieved the
2375 first row as an argument to ODBC_FETCH().
2376 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2377 dialplan. This function returns the content of the received message.
2378 * Added REPLACE, which searches a given variable name for a set of characters,
2379 then either replaces them with a single character or deletes them.
2380 * Added PASSTHRU, which literally passes the same argument back as its return
2381 value. The intent is to be able to use a literal string argument to
2382 functions that currently require a variable name as an argument.
2383 * HASH-associated variables now can be inherited across channel creation, by
2384 prefixing the name of the hash at assignment with the appropriate number of
2385 underscores, just like variables.
2386 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2387 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2388 whether or not channels that are bridged to the current channel will be
2389 required to have secure signaling and/or media.
2390 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2391 the current channel has secure signaling and/or media.
2392 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2393 "no_media_path" option.
2394 Returns "0" if there is a B channel associated with the call.
2395 Returns "1" if no B channel is associated with the call. The call is either
2396 on hold or is a call waiting call.
2397 * Added option to dialplan function CDR(), the 'f' option
2398 allows for high resolution times for billsec and duration fields.
2399 * FILE() now supports line-mode and writing.
2400 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2401 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2405 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2406 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2407 and is set when a dynamic feature is triggered.
2408 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2409 to dynamically create a new parking lot matching the value this varible is
2411 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2412 features.conf that should be the base for dynamic parkinglots.
2413 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2414 parkinglot should have.
2415 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2416 parkinglot should have.
2417 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2422 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2423 timeout has expired.
2424 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2425 to the caller when an Agent's phone is ringing. This can be used to indicate
2426 to the caller that their call is about to be picked up, which is nice when
2427 one has been on hold for an extened period of time.
2428 * A new config option, penaltymemberslimit, has been added to queues.conf.
2429 When set this option will disregard penalty settings when a queue has too
2431 * A new option, 'I' has been added to both app_queue and app_dial.
2432 By setting this option, Asterisk will not update the caller with
2433 connected line changes or redirecting party changes when they occur.
2434 * A 'relative-periodic-announce' option has been added to queues.conf. When
2435 enabled, this option will cause periodic announce times to be calculated
2436 from the end of announcements rather than from the beginning.
2437 * The autopause option in queues.conf can be passed a new value, "all." The
2438 result is that if a member becomes auto-paused, he will be paused in all
2439 queues for which he is a member, not just the queue that failed to reach
2441 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2442 * The queue logger now allows events to optionally propagate to a file,
2443 even when realtime logging is turned on. Additionally, realtime logging
2444 supports sending the event arguments to 5 individual fields, although it
2445 will fallback to the previous data definition, if the new table layout is
2448 mISDN channel driver (chan_misdn) changes
2449 ----------------------------------------
2450 * Added display_connected parameter to misdn.conf to put a display string
2451 in the CONNECT message containing the connected name and/or number if
2452 the presentation setting permits it.
2453 * Added display_setup parameter to misdn.conf to put a display string
2454 in the SETUP message containing the caller name and/or number if the
2455 presentation setting permits it.
2456 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2457 indicate the dialplan settings are to be obtained from the asterisk
2459 * Made misdn.conf parameter callerid accept the "name" <number> format
2460 used by the rest of the system.
2461 * Made use the nationalprefix and internationalprefix misdn.conf
2462 parameters to prefix any received number from the ISDN link if that
2463 number has the corresponding Type-Of-Number. NOTE: This includes
2464 comparing the incoming call's dialed number against the MSN list.
2465 * Added the following new parameters: unknownprefix, netspecificprefix,
2466 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2467 received number from the ISDN link if that number has the corresponding
2469 * Added new dialplan application misdn_command which permits controlling
2470 the CCBS/CCNR functionality.
2471 * Added new dialplan function mISDN_CC which permits retrieval of various
2472 values from an active call completion record.
2473 * For PTP, you should manually send the COLR of the redirected-to party
2474 for an incomming redirected call if the incoming call could experience
2475 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2476 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2477 if the REDIRECTING(from-num) is not empty.
2478 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2479 option on all of the REDIRECTING statements before dialing the
2480 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2481 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2482 redirecting-to presentation (COLR) when it becomes available.
2483 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2486 thirdparty mISDN enhancements
2487 -----------------------------
2488 mISDN has been modified by Digium, Inc. to greatly expand facility message
2490 * Enhanced COLP support for call diversion and transfer.
2491 * CCBS/CCNR support.
2493 The latest modified mISDN v1.1.x based version is available at:
2494 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2495 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2497 Tagged versions of the modified mISDN code are available under:
2498 http://svn.digium.com/svn/thirdparty/mISDN/tags
2499 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2501 libpri channel driver (chan_dahdi) DAHDI changes
2502 -------------------------------------------
2503 * The channel variable PRIREDIRECTREASON is now just a status variable
2504 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2505 to read and alter the reason.
2506 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2507 redirected-to party for an incomming redirected call if the incoming call
2508 could experience further redirects. Just set the
2509 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2510 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2512 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2513 use the inhibit(i) option on all of the REDIRECTING statements before
2514 dialing the redirected-to party. You still have to set the
2515 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2516 will update the redirecting-to presentation (COLR) when it becomes available.
2517 * Added the ability to ignore calls that are not in a Multiple Subscriber
2518 Number (MSN) list for PTMP CPE interfaces.
2519 * Added dynamic range compression support for dahdi channels. It is
2520 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2521 * Added support for ISDN calling and called subaddress with partial support
2522 for connected line subaddress.
2523 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2524 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2525 to transfer a held call on disconnect similar to an analog phone.
2526 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2527 Will reroute/deflect an outgoing call when receive the message.
2528 Can use the DAHDISendCallreroutingFacility to send the message for the
2530 * Added standard location to add options to chan_dahdi dialing:
2531 Dial(DAHDI/g1[/extension[/options]])
2534 R Reverse charging indication
2535 * Added Reverse Charging Indication (Collect calls) send/receive option.
2536 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2537 Dial(DAHDI/g1/extension/R)
2538 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2539 (requires latest LibPRI)
2540 * Added ability to send/receive keypad digits in the SETUP message.
2541 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2542 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2543 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2544 (requires latest LibPRI)
2545 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2546 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2547 back into the same interface. Tromboned calls happen because of call routing,
2548 call deflection, call forwarding, and call transfer.
2549 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2550 * Added the ability to support call waiting calls. (The SETUP has no B channel
2552 * Added Malicious Call ID (MCID) event to the AMI call event class.
2553 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2555 Asterisk Manager Interface
2556 --------------------------
2557 * The Hangup action now accepts a Cause header which may be used to
2558 set the channel's hangup cause.
2559 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2560 to specify a separate .pem file to hold a private key. By default sslcert
2561 is used to hold both the public and private key.
2562 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2563 for options containing the 'tls' prefix. For example, 'sslenable' is now
2564 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2565 across all .conf files. All affected sample.conf files have been modified to
2566 reflect this change. Previous options such as 'sslenable' still work,
2567 but options with the 'tls' prefix are preferred.
2568 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2569 in a channel. (res_mutestream.so)
2570 * The configuration file manager.conf now supports a channelvars option, which
2571 specifies a list of channel variables to include in each channel-oriented
2573 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2574 and ExtraPriority to allow redirecting the second channel to a different
2575 location than the first.
2576 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2578 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2579 in a MixMonitor recording.
2580 * The 'iax2 show peers' output is now similar to the expected output of
2582 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2584 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2585 AOC-E messages on a channel.
2586 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2587 conform more closely to similar events.
2588 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2590 * Added optional parkinglot variable for park command.
2591 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2592 if CallerIDNum and CallerIDName headers are also present.
2594 Channel Event Logging
2595 ---------------------
2596 * A new interface, CEL, is introduced here. CEL logs single events, much like
2597 the AMI, but it differs from the AMI in that it logs to db backends much
2598 like CDR does; is based on the event subsystem introduced by Russell, and
2599 can share in all its benefits; allows multiple backends to operate like CDR;
2600 is specialized to event data that would be of concern to billing sytems,
2601 like CDR. Backends for logging and accounting calls have been produced,
2602 but a new CDR backend is still in development.
2606 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2607 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2608 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2609 * Multiple files and formats can now be specified in cdr_custom.conf.
2610 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2611 See configs/cdr_syslog.conf.sample for more information.
2612 * A 'sequence' field has been added to CDRs which can be combined with
2613 linkedid or uniqueid to uniquely identify a CDR.
2614 * Handling of billsec and duration field has changed. If your table definition
2615 specifies those fields as float,double or similar they will now be logged with
2616 microsecond accuracy instead of a whole integer.
2618 Calendaring for Asterisk
2619 ------------------------
2620 * A new set of modules were added supporing calendar integration with Asterisk.
2621 Dialplan functions for reading from and writing to calendars are included,
2622 as well as the ability to execute dialplan logic upon calendar event notifications.
2623 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2624 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2625 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2626 2003 support does not support forms-based authentication).
2628 Call Completion Supplementary Services for Asterisk
2629 ---------------------------------------------------
2630 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2631 DAHDI/ISDN supports call completion for the following switch types:
2632 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2633 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2635 Multicast RTP Support
2636 ---------------------
2637 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2638 The channel driver can be used with the Page application to perform multicast RTP
2639 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2640 Type can be either basic or linksys.
2641 Destination is the IP address and port for the RTP packets.
2642 Control address is specific to the linksys type and is used for sending the control
2643 packets unique to them.
2645 Security Events Framework
2646 -------------------------
2647 * Asterisk has a new C API for reporting security events. The module res_security_log
2648 sends these events to the "security" logger level. Currently, AMI is the only
2649 Asterisk component that reports security events. However, SIP support will be
2650 coming soon. For more information on the security events framework, see the
2651 "Asterisk Security Framework" section of the Asterisk wiki at
2652 https://wiki.asterisk.org/wiki/x/wgBQ
2653 * SIP support was added in Asterisk 10
2654 * This API now supports IPv6 addresses
2658 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2659 * A spandsp based fax backend (res_fax_spandsp) has been added.
2660 * The app_fax module has been deprecated in favor of the res_fax module and
2661 the new res_fax_spandsp backend.
2662 * The SendFAX and ReceiveFAX applications now send their log messages to a
2663 'fax' logger level, instead of to the generic logger levels. To see these
2664 messages, the system's logger.conf file will need to direct the 'fax' logger
2665 level to one or more destinations; the logger.conf.sample file includes an
2666 example of how to do this. Note that if the 'fax' logger level is *not*
2667 directed to at least one destination, log messages generated by these
2668 applications will be lost, and that if the 'fax' logger level is directed to
2669 the console, the 'core set verbose' and 'core set debug' CLI commands will
2670 have no effect on whether the messages appear on the console or not.
2674 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2675 Now, in order to enable transmitting silence during record the transmit_silence
2676 option should be used. transmit_silence_during_record remains a valid option, but
2677 defaults to the behavior of the transmit_silence option.
2678 * Addition of the Unit Test Framework API for managing registration and execution
2679 of unit tests with the purpose of verifying the operation of C functions.
2680 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2681 XMPP text messages to the remote JID.
2682 * Modules.conf has a new option - "require" - that marks a module as critical for
2683 the execution of Asterisk.
2684 If one of the required modules fail to load, Asterisk will exit with a return
2686 * An 'X' option has been added to the asterisk application which enables #exec support.
2687 This allows #exec to be used in asterisk.conf.
2688 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2689 * A new lockconfdir option has been added to asterisk.conf to protect the
2690 configuration directory (/etc/asterisk by default) during reloads.
2691 * The parkeddynamic option has been added to features.conf to enable the creation
2692 of dynamic parkinglots.
2693 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2694 the reportalarms config option.
2695 * chan_dahdi supports dialing configuring and dialing by device file name.
2696 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2697 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2698 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2699 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2700 Handy for the above name-based syntax as it does not depend on
2701 initialization order.
2702 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2703 significant increase in performance (about 3X) for installations using this switchtype.
2704 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2705 AIS. For more information, please see the Distributed Device State section of the
2706 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2707 * The addition of G.719 pass-through support.
2708 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2709 during device configuration.
2710 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2711 have less than 3 lines on the LCD.
2712 * Realtime now supports database failover. See the sample extconfig.conf for details.
2713 * The addition of improved translation path building for wideband codecs. Sample
2714 rate changes during translation are now avoided unless absolutely necessary.
2715 * The addition of the res_stun_monitor module for monitoring and reacting to network
2716 changes while behind a NAT.
2717 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2718 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2719 These allow support for any Administration. Default is AT&T values.
2723 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2724 optionally accept a filename, to apply the setting only to the code generated from
2725 that source file when Asterisk was built. However, there are some modules in Asterisk
2726 that are composed of multiple source files, so this did not result in the behavior
2727 that users expected. In this version, 'core set debug' and 'core set verbose'
2728 can optionally accept *module* names instead (with or without the .so extension),
2729 which applies the setting to the entire module specified, regardless of which source
2730 files it was built from.
2731 * New 'manager show settings' command showing the current settings loaded from
2733 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2734 the channel hangup request to all channels.
2735 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2737 ------------------------------------------------------------------------------
2738 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2739 ------------------------------------------------------------------------------
2743 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2744 Snom phones use this for call pickup of extensions that the phone is
2746 * Added support for setting the domain in the URI for caller of an
2747 outbound call by using the SIPFROMDOMAIN channel variable.
2748 * Added a new configuration option "remotesecret" for authentication to
2749 remote services. For backwards compatibility, "secret" still has the
2750 same function as before, but now you can configure both a remote secret and a
2751 local secret for mutual authentication.
2752 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2753 the sound will be played to the target of an attended transfer
2754 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2755 finer control over how many peers Asterisk will qualify and the gap between them
2756 when all peers need to be qualified at the same time.
2757 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2758 (either globally or for a specific peer), chan_sip will treat any SDP data
2759 it receives as new data and update the media stream accordingly. By
2760 default, Asterisk will only modify the media stream if the SDP session
2761 version received is different from the current SDP session version. This
2762 option is required to interoperate with devices that have non-standard SDP
2763 session version implementations (observed with Microsoft OCS). This option
2764 is disabled by default.
2765 * The parsing of register => lines in sip.conf has been modified to allow a port
2766 to be present in the "user" portion. Please see the sip.conf.sample file for more
2768 * Added support for subscribing to MWI on a remote server and making the status available
2769 as a mailbox. Please see the sip.conf.sample file for more information.
2770 * Added a function to remove SIP headers added in the dialplan before the
2771 first INVITE is generated - SIPRemoveHeader()
2772 * Channel variables set with setvar= in a device configuration is now
2773 set both for inbound and outbound calls.
2774 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2778 * Added immediate option to iax.conf
2779 * Added forceencryption option to iax.conf
2780 * Added Encryption and Trunk status to manager command "iaxpeers"
2784 * The configuration file now holds separate sections for devices and lines.
2785 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2790 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2791 support for LibOpenR2. http://www.libopenr2.org/
2792 * The UK option waitfordialtone has been added for use with BT analog
2794 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2795 is used in conjunction with the 'faxdetect' configuration option. When
2796 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2797 switch to the configured faxbuffers policy. For example, to use 6 buffers
2798 and a 'full' buffer policy for a fax transmission, add:
2800 The faxbuffers configuration will be in affect until the call is torn down.
2801 * Added service message support for 4ESS/5ESS switches.
2805 * For DAHDI channels, the CHANNEL() dialplan function now
2806 supports changing the channel's buffer policy (for the current
2807 call only), using this syntax:
2809 exten => s,n,Set(CHANNEL(buffers)=6,full)
2811 This would change the channel to the 'full' buffer policy and
2812 6 (six) buffers. Possible options for this setting are the same
2813 as those in chan_dahdi.conf.
2814 * Added a new dialplan function, CURLOPT, which permits setting various
2815 options that may be useful with the CURL dialplan function, such as
2816 cookies, proxies, connection timeouts, passwords, etc.
2817 * Permit the syntax and synopsis fields of the corresponding dialplan
2818 functions to be individually set from func_odbc.conf.
2819 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2820 * func_odbc now may specify an insert query to execute, when the write query
2821 affects 0 rows (usually indicating that no such row exists).
2822 * Added a new dialplan function, LISTFILTER, which permits removing elements
2823 from a set list, by name. Uses the same general syntax as the existing CUT
2824 and FIELDQTY dialplan functions, which also manage lists.
2825 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2826 obtaining realtime data from the dialplan.
2827 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2828 a subroutine when using the GoSub() and Return() applications.
2829 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2830 of "core show function AUDIOHOOK_INHERIT" from the CLI
2831 * Added AES_ENCRYPT. For information on its use, please see the output
2832 of "core show function AES_ENCRYPT" from the CLI
2833 * Added AES_DECRYPT. For information on its use, please see the output
2834 of "core show function AES_DECRYPT" from the CLI
2835 * func_odbc now supports database transactions across multiple queries.
2839 * Scheduled meetme conferences may now have their end times extended by
2841 * app_authenticate now gives the ability to select a prompt other than
2843 * app_directory now pays attention to the searchcontexts setting in
2844 voicemail.conf and will look through all contexts, if no context is
2845 specified in the initial argument.
2846 * A new application, Originate, has been introduced, that allows asynchronous
2847 call origination from the dialplan.
2848 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2849 in addition to the setting in the "general" context.
2850 * Added ConfBridge dialplan application which does conference bridges without
2851 DAHDI. For information on its use, please see the output of
2852 "core show application ConfBridge" from the CLI.
2856 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2857 operation to the AMI Redirect action.
2858 * extensions.conf now allows you to use keyword "same" to define an extension
2859 without actually specifying an extension. It uses exactly the same pattern
2860 as previously used on the last "exten" line. For example:
2861 exten => 123,1,NoOp(something)
2862 same => n,SomethingElse()
2863 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2864 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2865 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2866 by the new clialiases module. See cli_aliases.conf.sample file.
2867 * Times within timespecs are now accurate down to the minute. This is a change
2868 from historical Asterisk, which only provided timespecs rounded to the nearest
2869 even (read: evenly divisible by 2) minute mark.
2870 * The realtime switch now supports an option flag, 'p', which disables searches for
2872 * In addition to a time range and date range, timespecs now accept a 5th optional
2873 argument, timezone. This allows you to perform time checks on alternate
2874 timezones, especially if those daylight savings time ranges vary from your
2875 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2877 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2878 give you the correct output for an asterisk box behind nat. It will give you the
2879 externhost and localnet settings.
2880 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2881 can connect calls in passthrough mode, as well as record and play back files.
2882 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2883 using pickupsound and pickupfailsound in features.conf.
2884 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2885 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2886 instead of the /var/run/asterisk.pid where it used to be. This will make
2887 installs as non-root easier to manage.
2892 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2893 be written; they will no longer be explicitly written.
2895 Asterisk Manager Interface
2896 --------------------------
2897 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2898 a non-empty value) in your request. If you do this, any pending AMI events will
2899 *not* be included in the response to your request as they would normally, but
2900 will be left in the event queue for the next request you make to retrieve. For
2901 some applications, this will allow you to guarantee that you will only see
2902 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2903 To know whether the Asterisk server supports this header or not, your client can
2904 inspect the first response back from the server to see if it includes this header:
2906 Pragma: SuppressEvents
2908 If this is included, the server supports event suppression.
2910 * Added 4 new Actions to list skinny device(s) and line(s)
2916 LDAP Schema File Additions
2917 --------------------------
2918 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2919 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2921 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2922 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2923 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2924 * Removed redundant IPaddr (there's already IPAddress)
2925 - Gives more configuration Flags for SIP-Users available (tested)
2926 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2927 without extensibleObject (which really should be the last resort); gives
2928 also additional possibilities for LDAP-filter
2930 ------------------------------------------------------------------------------
2931 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2932 ------------------------------------------------------------------------------
2934 Device State Handling
2935 ---------------------
2936 * The event infrastructure in Asterisk got another big update to help support
2937 distributed events. It currently supports distributed device state and
2938 distributed Voicemail MWI (Message Waiting Indication). A new module has
2939 been merged, res_ais, which facilitates communicating events between servers.
2940 It uses the SAForum AIS (Service Availability Forum Application Interface
2941 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2942 a cluster of Asterisk servers, and to share events between them. For more
2943 information on setting this up, refer to the Distributed Device State section
2944 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2948 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2949 variables from an Asterisk configuration file.
2950 * The JACK_HOOK function now has a c() option to supply a custom client name.
2951 * Added two new dialplan functions from libspeex for audio gain control and
2952 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2953 rx directions of a channel from the dialplan.
2954 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2955 based on other parameters. The default is still to search based on the
2956 forwarding station ID. However, there are new options that allow you to search
2957 based on the message desk terminal ID, or the message desk number.
2958 * TIMEOUT() has been modified to be accurate down to the millisecond.
2959 * ENUM*() functions now include the following new options:
2960 - 'u' returns the full URI and does not strip off the URI-scheme.
2961 - 's' triggers ISN specific rewriting
2962 - 'i' looks for branches into an Infrastructure ENUM tree
2963 - 'd' for a direct DNS lookup without any flipping of digits.
2964 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2965 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2966 deviation of jitter, rtt, and loss for a call using chan_sip.
2968 DAHDI channel driver (chan_dahdi) Changes
2969 ----------------------------------------
2970 * Channels can now be configured using named sections in chan_dahdi.conf, just
2971 like other channel drivers, including the use of templates.
2972 * The default for pridialplan has changed from 'national' to 'unknown'.
2976 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2977 to something that matches the pattern a hint will be created using the contents
2978 and variables evaluated.
2979 * Dialplan matching has been extended to allow an extension to return to the
2980 PBX core to wait for more digits. This is done by using the new dialplan
2981 application called "Incomplete". This will permit a whole new level of
2982 extension control, by giving the administrator more control over early
2983 matches employing one of the short-circuit pattern match operators. Note
2984 that custom applications can trigger this same behavior by returning the
2985 special value AST_PBX_INCOMPLETE.
2989 * Directory now permits both first and last names to be matched at the same
2990 time. In addition, the number of digits to enter of the name can be set in
2991 the arguments to Directory; previously, you could enter only 3, regardless
2992 of how many names are in your company. For large companies, this should be
2994 * Voicemail now permits a mailbox setting to wrap around from first to last
2995 messages, if the "messagewrap" option is set to a true value.
2996 * Voicemail now permits an external script to be run, for password validation.
2997 The script should output "VALID" or "INVALID" on stdout, depending upon the
2998 wish to validate or invalidate the password given. Arguments are:
2999 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3001 * Dial has a new option: F(context^extension^pri), which permits a callee to
3002 continue in the dialplan, at the specified label, if the caller hangs up.
3003 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3004 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3005 * The Jack application now has a c() option to supply a custom client name.
3006 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3007 like the pre-existing whisper mode, except that the spy can also talk to the
3008 participant on the bridged channel as well.
3009 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3010 to be spoken instead of the channel name or number. For more information on the
3011 use of this option, issue the command "core show application ChanSpy" from the
3013 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3014 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3015 words, if using the 'd' option, it is not possible to enter a number to append to
3016 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3017 change to whisper mode, and pressing 6 will change to barge mode.
3018 * ExternalIVR now takes several options that affect the way it performs, as
3019 well as having several new commands. Please see the External IVR page on the Asterisk
3020 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3021 * Added ability to communicate over a TCP socket instead of forking a child process for the
3022 ExternalIVR application.
3023 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3024 of just the first one if you give the function more then one channel to check.
3025 * PrivacyManager now takes an option where you can specify a context where the
3026 given number will be matched. This way you have more control over who is allowed
3027 and it stops the people who blindly enter 10 digits.
3028 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3029 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3030 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3031 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3032 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3033 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3034 * The Dial() application no longer copies the language used by the caller to the callee's
3035 channel. If you desire for the caller's channel's language to be used for file playback
3036 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3037 * SendImage() no longer hangs up the channel on error; instead, it sets the
3038 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3039 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3041 * Park has a new option, 's', which silences the announcement of the parking space number.
3042 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3043 invalid input and will be assumed to mean that no timeout is desired.
3047 * Added DNS manager support to registrations for peers referencing peer entries.
3048 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3049 as well as periodically updating the IP address. These properties allow for
3050 better performance as well as recovery in the event of an IP change.
3051 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3052 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3053 These changes also provide performance improvements for call setup and tear down.
3054 * Added ability to specify registration expiry time on a per registration basis in
3056 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3058 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3059 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3060 * 'sip show peers' and 'sip show users' display their entries sorted in
3061 alphabetical order, as opposed to the order they were in, in the config
3063 * Videosupport now supports an additional option, "always", which always sets
3064 up video RTP ports, even on clients that don't support it. This helps with
3065 callfiles and certain transfers to ensure that if two video phones are
3066 connected, they will always share video feeds.
3070 * Existing DNS manager lookups extended to check for SRV records.
3071 * IAX2 encryption support has been improved to support periodic key rotation
3072 within a call for enhanced security. The option "keyrotate" has been
3073 provided to disable this functionality to preserve backwards compatibility
3074 with older versions of IAX2 that do not support key rotation.
3078 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3079 data tree based on the given <path>.
3080 * New CLI command "data show providers" that will display all the registered
3082 * New CLI command, "config reload <file.conf>" which reloads any module that
3083 references that particular configuration file. Also added "config list"
3084 which shows which configuration files are in use.
3085 * New CLI commands, "pri show version" and "ss7 show version" that will
3086 display which version of libpri and libss7 are being used, respectively.
3087 A new API call was added so trunk will now have to be compiled against
3088 a versions of libpri and libss7 that have them or it will not know that
3089 these libraries exist.
3090 * The commands "core show globals", "core set global" and "core set chanvar" has
3091 been deprecated in favor of the more semanticly correct "dialplan show globals",
3092 "dialplan set chanvar" and "dialplan set global".
3093 * New CLI command "dialplan show chanvar" to list all variables associated
3094 with a given channel.
3098 * Addresses managed by DNS manager now can check to see if there is a DNS
3099 SRV record for a given domain and will use that hostname/port if present.
3101 AMI - The manager (TCP/TLS/HTTP)
3102 --------------------------------
3103 * The Status command now takes an optional list of variables to display
3104 along with channel status.
3105 * The QueueEntry event now also includes the channel's uniqueid
3109 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3110 as some people were running into this limit. This limit has been increased
3115 * The TRANSFER queue log entry now includes the the caller's original
3116 position in the transferred-from queue.
3117 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3118 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3119 as well as an explanation about timeout options in general
3120 * Added a new option - C - for forcing the "answered elsewhere" flag on
3121 cancellation of calls in to members of the queue. This is to avoid the
3122 call to a member of a queue having the call listed as a "missed call".
3126 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3127 adaptive capabilities. What this means in practical terms is that if your
3128 realtime table lacks critical fields, Asterisk will now emit warnings to
3129 that effect. Also, some of the realtime drivers have the ability (if
3130 configured) to automatically add those columns to the table with the
3131 correct type and length.
3135 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3136 the 'setvar' option to cause a given audio file to be played upon completion
3137 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3138 Skinny channels only.
3139 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3140 Hoard page on the Asterisk wiki for more information:
3141 https://wiki.asterisk.org/wiki/x/pQBB
3142 * Config file variables may now be appended to, by using the '+=' append
3143 operator. This is most helpful when working with long SQL queries in
3144 func_odbc.conf, as the queries no longer need to be specified on a single
3146 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3147 which will add a second to the billsec when the ending
3148 time is set, if the number in the microseconds field of the end time is
3149 greater than the number of microseconds in the answer time. This allows
3150 users to count the 'initiated' seconds in their billing records.
3152 ------------------------------------------------------------------------------
3153 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3154 ------------------------------------------------------------------------------
3156 AMI - The manager (TCP/TLS/HTTP)
3157 --------------------------------
3158 * Manager has undergone a lot of changes, all of them documented
3159 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3160 * Manager version has changed to 1.1
3161 * Added a new action 'CoreShowChannels' to list currently defined channels
3162 and some information about them.
3163 * Added a new action 'SIPshowregistry' to list SIP registrations.
3164 * Added TLS support for the manager interface and HTTP server
3165 * Added the URI redirect option for the built-in HTTP server
3166 * The output of CallerID in Manager events is now more consistent.
3167 CallerIDNum is used for number and CallerIDName for name.
3168 * Enable https support for builtin web server.
3169 See configs/http.conf.sample for details.
3170 * Added a new action, GetConfigJSON, which can return the contents of an
3171 Asterisk configuration file in JSON format. This is intended to help
3172 improve the performance of AJAX applications using the manager interface
3174 * SIP and IAX manager events now use "ChannelType" in all cases where we
3175 indicate channel driver. Previously, we used a mixture of "Channel"
3176 and "ChannelDriver" headers.
3177 * Added a "Bridge" action which allows you to bridge any two channels that
3178 are currently active on the system.
3179 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3180 the voicemail users setup.
3181 * Added 'DBDel' and 'DBDelTree' manager commands.
3182 * cdr_manager now reports events via the "cdr" level, separating it from
3183 the very verbose "call" level.
3184 * Manager users are now stored in memory. If you change the manager account
3185 list (delete or add accounts) you need to reload manager.
3186 * Added Masquerade manager event for when a masquerade happens between
3188 * Added "manager reload" command for the CLI
3189 * Lots of commands that only provided information are now allowed under the
3190 Reporting privilege, instead of only under Call or System.
3191 * The IAX* commands now require either System or Reporting privilege, to
3192 mirror the privileges of the SIP* commands.
3193 * Added ability to retrieve list of categories in a config file.
3194 * Added ability to retrieve the content of a particular category.
3195 * Added ability to empty a context.
3196 * Created new action to create a new file.
3197 * Updated delete action to allow deletion by line number with respect to category.
3198 * Added new action insert to add new variable to category at specified line.
3199 * Updated action newcat to allow new category to be inserted in file above another
3201 * Added new event "JitterBufStats" in the IAX2 channel
3202 * Originate now requires the Originate privilege and, if you want to call out
3203 to a subshell, it requires the System privilege, as well. This was done to
3204 enhance manager security.
3205 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3206 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3207 or manager show command Atxfer from the CLI
3208 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3209 details or manager show command IAXregistry from the CLI
3213 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3214 state in the dialplan, as well as creating custom device states that are
3215 controllable from the dialplan.
3216 * Extend CALLERID() function with "pres" and "ton" parameters to
3217 fetch string representation of calling number presentation indicator
3218 and numeric representation of type of calling number value.
3219 * MailboxExists converted to dialplan function
3220 * A new option to Dial() for telling IP phones not to count the call
3221 as "missed" when dial times out and cancels.
3222 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3223 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3224 held for any given channel. Also, locks are automatically freed when a
3226 * Added HINT() dialplan function that allows retrieving hint information.
3227 Hints are mappings between extensions and devices for the sake of
3228 determining the state of an extension. This function can retrieve the list
3229 of devices or the name associated with a hint.
3230 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3232 * Added SYSINFO() dialplan function which allows retrieval of system information
3233 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3234 the existence of a dialplan target.
3235 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3236 upper and lower case, respectively.
3237 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3238 ID for the call (not the Asterisk call ID or unique ID), provided that the
3239 channel driver supports this. For SIP, you get the SIP call-ID for the
3240 bridged channel which you can store in the CDR with a custom field.
3244 * Added CLI permissions, config file: cli_permissions.conf
3245 default is to allow all commands for every local user/group.
3246 Also this new feature added three new CLI commands:
3247 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3248 - cli reload permissions
3249 - cli show permissions
3250 * New CLI command "core show hint" (usage: core show hint <exten>)
3251 * New CLI command "core show settings"
3252 * Added 'core show channels count' CLI command.
3253 * Added the ability to set the core debug and verbose values on a per-file basis.
3254 * Added 'queue pause member' and 'queue unpause member' CLI commands
3255 * Ability to set process limits ("ulimit") without restarting Asterisk
3256 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3257 output to make debugging on busy systems much easier.
3258 * New CLI commands "dialplan set extenpatternmatching true/false"
3259 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3260 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3261 listed in the startup_commands section of cli.conf will get executed.
3262 * Added a CLI command, "devstate change", which allows you to set custom device
3263 states from the func_devstate module that provides the DEVICE_STATE() function
3264 and handling of the "Custom:" devices.
3265 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3266 sorted into the different possible callbacks, with the number of entries
3267 currently scheduled for each. Gives you a feel for how busy the sip channel
3269 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3270 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3271 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3275 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3276 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3277 for a received call. If it is detected, the channel will jump to the
3278 'fax' extension in the dialplan.
3279 * The default SIP useragent= identifier now includes the Asterisk version
3280 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3281 If set, and the incoming request carries authentication info,
3282 the username to match in the users list is taken from the Digest header
3283 rather than from the From: field. This feature is considered experimental.
3284 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3285 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3286 * The "localmask" setting was removed in version 1.2 and the reminder about it
3287 being removed is now also removed.
3288 * A new option "busylevel" for setting a level of calls where asterisk reports
3289 a device as busy, to separate it from call-limit. This value is also added
3290 to the SIP_PEER dialplan function.
3291 * A new realtime family called "sipregs" is now supported to store SIP registration
3292 data. If this family is defined, "sippeers" will be used for configuration and
3293 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3294 registration data, as before.
3295 * The SIPPEER function have new options for port address, call and pickup groups
3296 * Added support for T.140 realtime text in SIP/RTP
3297 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3298 required due to the restructuring of how MWI is handled. See the descriptions
3299 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3300 for more information.
3301 * Added rtpdest option to CHANNEL() dialplan function.
3302 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3303 * SIP now adds a header to the CANCEL if the call was answered by another phone
3304 in the same dial command, or if the new c option in dial() is used.
3305 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3306 states it is not needed. For phones, however, that do require it the "registertrying" option
3307 has been added so it can be enabled.
3308 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3309 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3310 used to enable this functionality).
3311 * New settings for timer T1 and timer B on a global level or per device. This makes it
3312 possible to force timeout faster on non-responsive SIP servers. These settings are
3313 considered advanced, so don't use them unless you have a problem.
3314 * Added a dial string option to be able to set the To: header in an INVITE to any
3316 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3317 the qualify frequency.
3318 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3319 were not properly torn down due to network or endpoint failures during an established
3321 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3322 and configs/sip.conf.sample for more information on how it is used.
3323 * Added a new configuration option "authfailureevents" that enables manager events when
3324 a peer can't authenticate properly.
3325 * Added DNS manager support to registrations for peers not referencing a peer entry.
3329 * Added the trunkmaxsize configuration option to chan_iax2.
3330 * Added the srvlookup option to iax.conf
3331 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3334 XMPP Google Talk/Jingle changes
3335 -------------------------------
3336 * Added the bindaddr option to gtalk.conf.
3340 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
3341 * Proper codec support in chan_skinny.
3342 * Added settings for IP and Ethernet QoS requests
3346 * Added separate settings for media QoS in mgcp.conf
3348 Console Channel Driver changes
3349 ------------------------------
3350 * Added experimental support for video send & receive to chan_oss.
3351 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
3354 Phone channel changes (chan_phone)
3355 ----------------------------------
3356 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
3358 H.323 channel Changes
3359 ---------------------
3360 * H323 remote hold notification support added (by NOTIFY message
3361 and/or H.450 supplementary service)
3363 Local channel changes
3364 ---------------------
3365 * The device state functionality in the Local channel driver has been updated
3366 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
3367 to just UNKNOWN if the extension exists.
3368 * Added jitterbuffer support for chan_local. This allows you to use the
3369 generic jitterbuffer on incoming calls going to Asterisk applications.
3370 For example, this would allow you to use a jitterbuffer for an incoming
3371 SIP call to Voicemail by putting a Local channel in the middle. This
3372 feature is enabled by using the 'j' option in the Dial string to the Local
3373 channel in conjunction with the existing 'n' option for local channels.
3374 * A 'b' option has been added which causes chan_local to return the actual channel
3375 that is behind it when queried. This is useful for transfer scenarios as the
3376 actual channel will be transferred, not the Local channel.
3378 Agent channel changes
3379 ----------------------
3380 * The ackcall and endcall options are now supplemented with options acceptdtmf
3381 and enddtmf. These allow for the DTMF keypress to be configurable. The options
3382 default to their old hard-coded values ('#' and '*' respectively) so this should
3383 not break any existing agent installations.
3385 DAHDI channel driver (chan_dahdi) Changes
3386 ----------------------------------------
3387 * SS7 support (via libss7 library)
3388 * In India, some carriers transmit CID via dtmf. Some code has been added
3389 that will handle some situations. The cidstart=polarity_IN choice has been added for
3390 those carriers that transmit CID via dtmf after a polarity change.
3391 * CID matching information is now shown when doing 'dialplan show'.
3392 * Added dahdi show version CLI command.
3393 * Added setvar support to chan_dahdi.conf channel entries.
3394 * Added two new options: mwimonitor and mwimonitornotify. These options allow
3395 you to enable MWI monitoring on FXO lines. When the MWI state changes,
3396 the script specified in the mwimonitornotify option is executed. An internal
3397 event indicating the new state of the mailbox is also generated, so that
3398 the normal MWI facilities in Asterisk work as usual.
3399 * Added signalling type 'auto', which attempts to use the same signalling type
3400 for a channel as configured in DAHDI. This is primarily designed for analog
3401 ports, but will also work for digital ports that are configured for FXS or FXO
3402 signalling types. This mode is also the default now, so if your chan_dahdi.conf
3403 does not specify signalling for a channel (which is unlikely as the sample
3404 configuration file has always recommended specifying it for every channel) then
3405 the 'auto' mode will be used for that channel if possible.
3406 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
3407 state for a channel; also ensured that the DNDState Manager event is
3408 emitted no matter how the DND state is set or cleared.
3412 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
3413 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
3414 for details. This new channel driver allows you to use Nortel i2002,
3415 i2004, and i2050 phones with Asterisk.
3416 * Added a new channel driver, chan_console, which uses portaudio as a cross
3417 platform audio interface. It was written as a channel driver that would
3418 work with Mac CoreAudio, but portaudio supports a number of other audio
3419 interfaces, as well. Note that this channel driver requires v19 or higher
3420 of portaudio; older versions have a different API.
3424 * Added the ability to specify arguments to the Dial application when using
3425 the DUNDi switch in the dialplan.
3426 * Added the ability to set weights for responses dynamically. This can be
3427 done using a global variable or a dialplan function. Using the SHELL()
3428 function would allow you to have an external script set the weight for
3430 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
3431 functions will allow you to initiate a DUNDi query from the dialplan,
3432 find out how many results there are, and access each one.
3433 * Added the ability to specifiy a port for a dundi peer.
3437 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
3438 functions will allow you to initiate an ENUM lookup from the dialplan,
3439 and Asterisk will cache the results. ENUMRESULT can be used to access
3440 the results without doing multiple DNS queries.
3444 * Added the ability to customize which sound files are used for some of the
3445 prompts within the Voicemail application by changing them in voicemail.conf
3446 * Added the ability for the "voicemail show users" CLI command to show users
projects / asterisk/asterisk.git / blob