1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
13 ------------------------------------------------------------------------------
16 --------------------------
17 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
18 conference user menus.
20 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
21 menus, bridge settings, and user settings that have been applied by the
22 CONFBRIDGE dialplan function.
24 * Added conference user option 'announce_join_leave_review'. This option
25 implies 'announce_join_leave' with the added effect that the user will
26 be asked if they want to confirm or re-record the recording of their
27 name when entering the conference
30 --------------------------
31 * At exit, the Directory application now sets a channel variable
32 DIRECTORY_RESULT to one of the following based on the reason for exiting:
33 OPERATOR user requested operator by pressing '0' for operator
34 ASSISTANT user requested assistant by pressing '*' for assistant
35 TIMEOUT user pressed nothing and Directory stopped waiting
36 HANGUP user's channel hung up
37 SELECTED user selected a user from the directory and is routed
38 USEREXIT user pressed '#' from the selection prompt to exit
39 FAILED directory failed in a way that wasn't accounted for. Dang.
42 --------------------------
43 * PickupChan now accepts channel uniqueids of channels to pickup.
46 --------------------------
47 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
48 to 'true' (case insensitive), then any Say application (SayNumber,
49 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
50 anticipate DTMF. If DTMF is received, these applications will behave like
51 the background application and jump to the received extension once a match
52 is established or after a short period of inactivity.
54 ------------------------------------------------------------------------------
55 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
56 ------------------------------------------------------------------------------
61 Asterisk 12 is a standard release of the Asterisk project. As such, the
62 focus of development for this release was on core architectural changes and
63 major new features. This includes:
64 * A more flexible bridging core based on the Bridging API
65 * A new internal message bus, Stasis
66 * Major standardization and consistency improvements to AMI
67 * Addition of the Asterisk RESTful Interface (ARI)
68 * A new SIP channel driver, chan_pjsip
69 In addition, as the vast majority of bridging in Asterisk was migrated to the
70 Bridging API used by ConfBridge, major changes were made to most of the
71 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
73 Specifications have been written for the affected interfaces. These
74 specifications are available on the Asterisk wiki:
75 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
76 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
77 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
79 It is *highly* recommended that anyone migrating to Asterisk 12 read the
80 information regarding its release both in this file and in the accompanying
81 UPGRADE.txt file. More detailed information on the major changes can be found
82 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
87 * Added build option DISABLE_INLINE. This option can be used to work around a
88 bug in gcc. For more information, see
89 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
91 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
92 the CHANNEL_TRACE build option were incompatible with the new bridging
95 * Asterisk now optionally uses libxslt to improve XML documentation generation
96 and maintainability. If libxslt is not available on the system, some XML
97 documentation will be incomplete.
99 * Asterisk now depends on libjansson. If a package of libjansson is not
100 available on your distro, please see http://www.digip.org/jansson/.
102 * Asterisk now depends on libuuid and, optionally, uriparser. It is
103 recommended that you install uriparser, even if it is optional.
105 * The new SIP stack and channel driver uses a particular version of PJSIP.
106 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
107 configuring and installing PJSIP for usage with Asterisk.
109 * Optional API was re-implemented to be more portable, and no longer requires
110 weak reference support from the compiler. The build option OPTIONAL_API may
111 be disabled to disable Optional API support.
118 * Along with AgentRequest, this application has been modified to be a
119 replacement for chan_agent. The act of a channel calling the AgentLogin
120 application places the channel into a pool of agents that can be
121 requested by the AgentRequest application. Note that this application, as
122 well as all other agent related functionality, is now provided by the
123 app_agent_pool module. See chan_agent and AgentRequest for more information.
125 * This application no longer performs agent authentication. If authentication
126 is desired, the dialplan needs to perform this function using the
127 Authenticate or VMAuthenticate application or through an AGI script before
130 * If this application is called and the agent is already logged in, the
131 dialplan will continue exection with the AGENT_STATUS channel variable set
132 to ALREADY_LOGGED_IN.
134 * The agents.conf schema has changed. Rather than specifying agents on a
135 single line in comma delineated fashion, each agent is defined in a separate
136 context. This allows agents to use the power of context templates in their
139 * A number of parameters from agents.conf have been removed. This includes
140 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
141 urlprefix, and savecallsin. These options were obsoleted by the move from
142 a channel driver model to the bridging/application model provided by
147 * A new application, this will request a logged in agent from the pool and
148 bridge the requested channel with the channel calling this application.
149 Logged in agents are those channels that called the AgentLogin application.
150 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
151 application will be set with an appropriate error value.
155 * This application has been removed. It was a holdover from when
156 AgentCallbackLogin was removed.
160 * Added support for additional Ademco DTMF signalling formats, including
161 Express 4+1, Express 4+2, High Speed and Super Fast.
163 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
164 call time, in milliseconds, to run the application.
166 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
167 maximum number of times to retry the call.
169 * Added a new configuration option answait. If set, the AlarmReceiver
170 application will wait the number of milliseconds specified by answait
171 after the channel has answered. Valid values range between 500
172 milliseconds and 10000 milliseconds.
174 * Added configuration option no_group_meta. If enabled, grouping of metadata
175 information in the AlarmReceiver log file will be skipped.
179 * A new application in Asterisk, this will place the calling channel
180 into a holding bridge, optionally entertaining them with some form of
181 media. Channels participating in a holding bridge do not interact with
182 other channels in the same holding bridge. Optionally, however, a channel
183 may join as an announcer. Any media passed from an announcer channel is
184 played to all channels in the holding bridge. Channels leave a holding
185 bridge either when an optional timer expires, or via the ChannelRedirect
186 application or AMI Redirect action.
190 * All participants in a bridge can now be kicked out of a conference room
191 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
192 command, i.e., 'confbridge kick <conference> all'
194 * CLI output for the 'confbridge list' command has been improved. When
195 displaying information about a particular bridge, flags will now be shown
196 for the participating users indicating properties of that user.
198 * The ConfbridgeList event now contains the following fields: WaitMarked,
199 EndMarked, and Waiting. This displays additional properties about the
200 user's profile, as well as whether or not the user is waiting for a
201 Marked user to enter the conference.
203 * Added a new option for conference recording, record_file_append. If enabled,
204 when the recording is stopped and then re-started, the existing recording
205 will be used and appended to.
207 * ConfBridge now has the ability to set the language of announcements to the
208 conference. The language can be set on a bridge profile in confbridge.conf
209 or by the dialplan function CONFBRIDGE(bridge,language)=en.
213 * The channel variable CPLAYBACKSTATUS may now return the value
214 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
215 such as AMI. See the AMI action ControlPlayback for more information.
219 * Added the 'a' option, which allows the caller to enter in an additional
220 alias for the user in the directory. This option must be used in conjunction
221 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
222 specified in voicemail.conf.
226 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
227 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
228 containing the unique ID of the bridge that the channel happens to be in.
232 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
233 for more information.
235 * Variables are no longer purged from the original CDR. See the 'v' option for
238 * The 'A' option has been removed. The Answer time on a CDR is never updated
241 * The 'd' option has been removed. The disposition on a CDR is a function of
242 the state of the channel and cannot be altered.
244 * The 'D' option has been removed. Who the Party B is on a CDR is a function
245 of the state of the respective channels involved in the CDR and cannot be
248 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
249 such that the start time and, if applicable, the answer time was updated.
250 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
251 'r' option now triggers the Reset, setting the start time (and answer time
252 if applicable) to the current time. Note that the 'a' option still sets
253 the answer time to the current time if the channel was already answered.
255 * The 's' option has been removed. A variable can be set on the original CDR
256 if desired using the CDR function, and removed from a forked CDR using the
259 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
260 longer applies in the CDR engine.
262 * The 'v' option now prevents the copy of the variables from the original CDR
263 to the forked CDR. Previously the variables were always copied but were
264 removed from the original. This was changed as removing variables from a CDR
265 can have unintended side effects - this option allows the user to prevent
266 propagation of variables from the original to the forked without modifying
271 * Added the 'n' option to MeetMe to prevent application of the DENOISE
272 function to a channel joining a conference. Some channel drivers that vary
273 the number of audio samples in a voice frame will experience significant
274 quality problems if a denoiser is attached to the channel; this option gives
275 them the ability to remove the denoiser without having to unload func_speex.
279 * The 'b' option now includes conferences as well as sounds played to the
282 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
283 running during a transfer. If a MixMonitor is started on a channel,
284 the MixMonitor will continue to record the audio passing through the
285 channel even in the presence of transfers.
289 * The NoCDR application is deprecated. Please use the CDR_PROP function to
292 * While the NoCDR application will prevent CDRs for a channel from being
293 propagated to registered CDR backends, it will not prevent that data from
294 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
295 function that enables CDRs on a channel will restore those records that have
296 not yet been finalized.
300 * The app_parkandannounce module has been removed. The application
301 ParkAndAnnounce is now provided by the res_parking module. See the
302 res_parking changes for more information.
306 * Added queue available hint. The hint can be added to the dialplan using the
307 following syntax: exten,hint,Queue:{queue_name}_avail
308 For example, if the name of the queue is 'markq':
309 exten => 8501,hint,Queue:markq_avail
310 This will report 'InUse' if there are no logged in agents or no free agents.
311 It will report 'Idle' when an agent is free.
313 * Queues now support a hint for member paused state. The hint uses the form
314 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
315 are the name of the queue and the name of the member to subscribe to,
316 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
317 Members will show as In Use when paused.
319 * The configuration options eventwhencalled and eventmemberstatus have been
320 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
321 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
322 sent. The "Variable" fields will also no longer exist on the Agent* events.
323 These events can be filtered out from a connected AMI client using the
324 eventfilter setting in manager.conf.
326 * The queue log now differentiates between blind and attended transfers. A
327 blind transfer will result in a BLINDTRANSFER message with the destination
328 context and extension. An attended transfer will result in an
329 ATTENDEDTRANSFER message. This message will indicate the method by which
330 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
331 for running an application on a bridge or channel, or "LINK" for linking
332 two bridges together with local channels. The queue log will also now detect
333 externally initiated blind and attended transfers and record the transfer
336 * When performing queue pause/unpause on an interface without specifying an
337 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
338 least one member of any queue exists for that interface.
340 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
341 for realtime queue log entries.
345 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
346 CDRs when they were previously disabled on a channel.
348 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
349 backends occurs on an as-needed basis in order to preserve linkedid
350 propagation and other needed behavior.
354 * A new application, this is similar to SayAlpha except that it supports
355 case sensitive playback of the specified characters. For example,
356 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
360 * This application is deprecated in favor of CHANNEL(amaflags).
364 * The SendDTMF application will now accept 'W' as valid input. This will cause
365 the application to delay one second while streaming DTMF.
369 * A new application in Asterisk 12, this hands control of the channel calling
370 the application over to an external system. Currently, external systems
371 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
375 * UserEvent will now handle duplicate keys by overwriting the previous value
378 * In addition to AMI, UserEvent invocations will now be distributed to any
379 interested Stasis applications.
383 * The voicemail.conf configuration file now has an 'alias' configuration
384 parameter for use with the Directory application. The voicemail realtime
385 database table schema has also been updated with an 'alias' column.
390 * Pass through support has been added for both VP8 and Opus.
392 * Added format attribute negotiation for the Opus codec. Format attribute
393 negotiation is provided by the res_format_attr_opus module.
398 * Masquerades as an operation inside Asterisk have been effectively hidden
399 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
400 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
401 dropping of frame/audio hooks, and other internal implementation details
402 that users had to deal with. This fundamental change has large implications
403 throughout the changes documented for this version. For more information
404 about the new core architecture of Asterisk, please see the Asterisk wiki.
406 * Multiple parties in a bridge may now be transferred. If a participant in a
407 multi-party bridge initiates a blind transfer, a Local channel will be used
408 to execute the dialplan location that the transferer sent the parties to. If
409 a participant in a multi-party bridge initiates an attended transfer,
410 several options are possible. If the attended transfer results in a transfer
411 to an application, a Local channel is used. If the attended transfer results
412 in a transfer to another channel, the resulting channels will be merged into
415 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
416 driver specific. If the channel variable is set on the transferrer channel,
417 the sound will be played to the target of an attended transfer.
419 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
420 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
421 listed. Any more peers in the bridge will not be included in the list.
422 BRIDGEPEER is not valid in holding bridges like parking since those channels
423 do not talk to each other even though they are in a bridge.
425 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
426 and will contain a value if the BRIDGEPEER's channel driver supports it.
428 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
429 was responsible for an attended transfer in a similar fashion to
432 * Modules using the Configuration Framework or Sorcery must have XML
433 configuration documentation. This configuration documentation is included
434 with the rest of Asterisk's XML documentation, and is accessible via CLI
435 commands. See the CLI changes for more information.
437 AMI (Asterisk Manager Interface)
439 * Major changes were made to both the syntax as well as the semantics of the
440 AMI protocol. In particular, AMI events have been substantially improved
441 in this version of Asterisk. For more information, please see the AMI
442 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
444 * AMI events that reference a particular channel or bridge will now always
445 contain a standard set of fields. When multiple channels or bridges are
446 referenced in an event, fields for at least some subset of the channels
447 and bridges in the event will be prefixed with a descriptive name to avoid
448 name collisions. See the AMI event documentation on the Asterisk wiki for
451 * The CLI command 'manager show commands' no longer truncates command names
452 longer than 15 characters and no longer shows authorization requirement
453 for commands. 'manager show command' now displays the privileges needed
454 for using a given manager command instead.
456 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
457 peer in its response if the peer has a subscribe context set.
459 * The SIPqualifypeer action now acknowledges the request once it has
460 established that the request is against a known peer. It also issues a new
461 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
463 * The PlayDTMF action now supports an optional 'Duration' parameter. This
464 specifies the duration of the digit to be played, in milliseconds.
466 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
467 updates when changes occur instead of requiring the use of pollmailboxes.
469 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
470 AMI client to manipulate audio currently being played back on a channel. The
471 supported operations depend on the application being used to send audio to
472 the channel. When the audio playback was initiated using the ControlPlayback
473 application or CONTROL STREAM FILE AGI command, the audio can be paused,
474 stopped, restarted, reversed, or skipped forward. When initiated by other
475 mechanisms (such as the Playback application), the audio can be stopped,
476 reversed, or skipped forward.
478 * Channel related events now contain a snapshot of channel state, adding new
479 fields to many of these events.
481 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
482 in a future release. Please use the common 'Exten' field instead.
484 * The AMI event 'UserEvent' from app_userevent now contains the channel state
485 fields. The channel state fields will come before the body fields.
487 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
488 'UnParkedCall' have changed significantly in the new res_parking module.
490 The 'Channel' and 'From' headers are gone. For the channel that was parked
491 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
492 has a number of fields associated with it. The old 'Channel' header relayed
493 the same data as the new 'ParkeeChannel' header.
495 The 'From' field was ambiguous and changed meaning depending on the event.
496 for most of these, it was the name of the channel that parked the call
497 (the 'Parker'). There is no longer a header that provides this channel name,
498 however the 'ParkerDialString' will contain a dialstring to redial the
499 device that parked the call.
501 On UnParkedCall events, the 'From' header would instead represent the
502 channel responsible for retrieving the parkee. It receives a channel
503 snapshot labeled 'Retriever'. The 'from' field is is replaced with
506 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
508 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
509 fashion has changed the field names 'StartExten' and 'StopExten' to
510 'StartSpace' and 'StopSpace' respectively.
512 * The deprecated use of | (pipe) as a separator in the channelvars setting in
513 manager.conf has been removed.
515 * Channel Variables conveyed with a channel no longer contain the name of the
516 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
517 ChanVariable: bar=baz. When multiple channels are present in a single AMI
518 event, the various ChanVariable fields will contain a suffix that specifies
519 which channel they correspond to.
521 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
522 event always conveys the AMI event for a particular channel.
524 * All 'Reload' events have been consolidated into a single event type. This
525 event will always contain a Module field specifying the name of the module
526 and a Status field denoting the result of the reload. All modules now issue
527 this event when being reloaded.
529 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
530 fail to receive this event due to being connected after modules have loaded.
531 AMI connections that want to know when Asterisk is ready should listen for
532 the 'FullyBooted' event.
534 * app_fax now sends the same send fax/receive fax events as res_fax. The
535 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
536 now the 'ReceiveFAX' event.
538 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
539 'MusicOnHoldStop'. The sub type field has been removed.
541 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
542 carrier for another protocol.
544 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
545 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
546 to the specific channel. 'Both' may be specified to play a tone to both
547 channels. The old 'yes' option is still accepted as a way of playing the
548 tone to Channel2 only.
550 * The AMI 'Status' response event to the AMI Status action replaces the
551 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
552 indicate what bridge the channel is currently in.
554 * The AMI 'Hold' event has been moved out of individual channel drivers, into
555 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
558 * The AMI events in app_queue have been made more consistent with each other.
559 Events that reference channels (QueueCaller* and Agent*) will show
560 information about each channel. The (infamous) 'Join' and 'Leave' AMI
561 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
563 * The 'MCID' AMI event now publishes a channel snapshot when available and
564 its non-channel-snapshot parameters now use either the "MCallerID" or
565 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
566 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
567 parameters in the channel snapshot.
569 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
570 'AgentLogin' and 'AgentLogoff' respectively.
572 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
573 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
575 * 'ChannelUpdate' events have been removed.
577 * All AMI events now contain a 'SystemName' field, if available.
579 * Local channel optimization is now conveyed in two events:
580 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
581 when the Local channel driver begins attempting to optimize itself out of
582 the media path; the End event is sent after the channel halves have
583 successfully optimized themselves out of the media path.
585 * Local channel information in events is now prefixed with 'LocalOne' and
586 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
587 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
588 and 'LocalOptimizationEnd' events.
590 * The option 'allowmultiplelogin' can now be set or overriden in a particular
591 account. When set in the general context, it will act as the default
592 setting for defined accounts.
594 * The 'BridgeAction' event was removed. It technically added no value, as the
595 Bridge Action already receives confirmation of the bridge through a
596 successful completion Event.
598 * The 'BridgeExec' events were removed. These events duplicated the events that
599 occur in the Briding API, and are conveyed now through BridgeCreate,
600 BridgeEnter, and BridgeLeave events.
602 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
603 previous versions. They now report all SR/RR packets sent/received, and
604 have been restructured to better reflect the data sent in a SR/RR. In
605 particular, the event structure now supports multiple report blocks.
607 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
608 raised when a blind transfer/attended transfer completes successfully.
609 They contain information about the transfer that just completed, including
610 the location of the transfered channel.
612 * Added a 'security' class to AMI which outputs the required fields for
613 security messages similar to the log messages from res_security_log
615 CDR (Call Detail Records)
617 * Significant changes have been made to the behavior of CDRs. The CDR engine
618 was effectively rewritten and built on the Stasis message bus. For a full
619 definition of CDR behavior in Asterisk 12, please read the specification
620 on the Asterisk wiki (wiki.asterisk.org).
622 * CDRs will now be created between all participants in a bridge. For each
623 pair of channels in a bridge, a CDR is created to represent the path of
624 communication between those two endpoints. This lets an end user choose who
625 to bill for what during bridge operations with multiple parties.
627 * The duration, billsec, start, answer, and end times now reflect the times
628 associated with the current CDR for the channel, as opposed to a cumulative
629 measurement of all CDRs for that channel.
631 * When a CDR is dispatched, user defined CDR variables from both parties are
632 included in the resulting CDR. If both parties have the same variable, only
633 the Party A value is provided.
635 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
636 information regarding the CDR engine is logged as verbose messages. This
637 option should only be used if the behavior of the CDR engine needs to be
640 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
641 normally configured in cdr.conf.
643 * Added CLI command 'cdr show active {channel}'. When {channel} is not
644 specified, this command provides a summary of the channels with CDR
645 information and their statistics. When {channel} is specified, it shows
646 detailed information about all records associated with {channel}.
648 CEL (Channel Event Logging)
650 * CEL has undergone significant rework in Asterisk 12, and is now built on the
651 Stasis message bus. Please see the specification for CEL on the Asterisk
652 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
655 * The 'extra' field of all CEL events that use it now consists of a JSON blob
656 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
658 * BLINDTRANSFER events now report the transferee bridge unique
659 identifier, extension, and context in a JSON blob as the extra string
660 instead of the transferee channel name as the peer.
662 * ATTENDEDTRANSFER events now report the peer as NULL and additional
663 information in the 'extra' string as a JSON blob. For transfers that occur
664 between two bridged channels, the 'extra' JSON blob contains the primary
665 bridge unique identifier, the secondary channel name, and the secondary
666 bridge unique identifier. For transfers that occur between a bridged channel
667 and a channel running an app, the 'extra' JSON blob contains the primary
668 bridge unique identifier, the secondary channel name, and the app name.
670 * LOCAL_OPTIMIZE events have been added to convey local channel
671 optimizations with the record occurring for the semi-one channel and
672 the semi-two channel name in the peer field.
674 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
675 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
676 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
677 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
678 regardless of whether or not that bridge happens to contain multiple
683 * When compiled with '--enable-dev-mode', the astobj2 library will now add
684 several CLI commands that allow for inspection of ao2 containers that
685 register themselves with astobj2. The CLI commands are 'astobj2 container
686 dump', 'astobj2 container stats', and 'astobj2 container check'.
688 * Added specific CLI commands for bridge inspection. This includes 'bridge
689 show all', which lists all bridges in the system, and 'bridge show {id}',
690 which provides specific information about a bridge.
692 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
693 ejecting the channels currently in the bridge. If the channels cannot
694 continue in the dialplan or application that put them in the bridge, they
697 * Added command 'bridge kick'. This will eject a single channel from a bridge.
699 * Added commands to inspect and manipulate the registered bridge technologies.
700 This include 'bridge technology show', which lists the registered bridge
701 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
702 which controls whether or not a registered bridge technology can be used
703 during smart bridge operations. If a technology is suspended, it will not
704 be used when a bridge technology is picked for channels; when unsuspended,
705 it can be used again.
707 * The command 'config show help {module} {type} {option}' will show
708 configuration documentation for modules with XML configuration
709 documentation. When {module}, {type}, and {option} are omitted, a listing
710 of all modules with registered documentation is displayed. When {module}
711 is specified, a listing of all configuration types for that module is
712 displayed, along with their synopsis. When {module} and {type} are
713 specified, a listing of all configuration options for that type are
714 displayed along with their synopsis. When {module}, {type}, and {option}
715 are specified, detailed information for that configuration option is
718 * Added 'core show sounds' and 'core show sound' CLI commands. These display
719 a listing of all installed media sounds available on the system and
720 detailed information about a sound, respectively.
722 * 'xmldoc dump' has been added. This CLI command will dump the XML
723 documentation DOM as a string to the specified file. The Asterisk core
724 will populate certain XML elements pulled from the source files with
725 additional run-time information; this command lets a user produce the
726 XML documentation with all information.
730 * Parking has been pulled from core and placed into a separate module called
731 res_parking. See Parking changes below for more details. Configuration for
732 parking should now be performed in res_parking.conf. Configuration for
733 parking in features.conf is now unsupported.
735 * Core attended transfers now have several new options. While performing an
736 attended transfer, the transferer now has the following options:
737 - *1 - cancel the attended transfer (configurable via atxferabort)
738 - *2 - complete the attended transfer, dropping out of the call
739 (configurable via atxfercomplete)
740 - *3 - complete the attended transfer, but stay in the call. This will turn
741 the call into a multi-party bridge (configurable via atxferthreeway)
742 - *4 - swap to the other party. Once an attended transfer has begun, this
743 options may be used multiple times (configurable via atxferswap)
745 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
746 must be on the channel initiating the transfer to have any effect.
748 * The BRIDGE_FEATURES channel variable would previously only set features for
749 the calling party and would set this feature regardless of whether the
750 feature was in caps or in lowercase. Use of a caps feature for a letter
751 will now apply the feature to the calling party while use of a lowercase
752 letter will apply that feature to the called party.
754 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
756 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
757 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
758 activated the dynamic feature.
760 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
761 only on the channel executing the dynamic feature. Executing a dynamic
762 feature on the bridge peer in a multi-party bridge will execute it on all
763 peers of the activating channel.
765 * You can now have the settings for a channel updated using the FEATURE()
766 and FEATUREMAP() functions inherited to child channels by setting
767 FEATURE(inherit)=yes.
769 * automixmon now supports additional channel variables from automon including:
770 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
771 and TOUCH_MIXMONITOR_MESSAGE_STOP
773 * A new general features.conf option 'recordingfailsound' has been added which
774 allowssetting a failure sound for a user tries to invoke a recording feature
775 such as automon or automixmon and it fails.
777 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
778 features.c for atxferdropcall=no to work properly. This option now just
783 * Added log rotation strategy 'none'. If set, no log rotation strategy will
784 be used. Given that this can cause the Asterisk log files to grow quickly,
785 this option should only be used if an external mechanism for log management
790 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
791 will store the path information for that peer when it registers. Realtime
792 tables can also use the 'supportpath' field to enable Path header support.
794 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
795 objectIdentifier. This maps to the supportpath option in sip.conf.
799 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
800 provides modules a useful abstraction on top of the many storage mechanisms
801 in Asterisk, including the Asterisk Database, static configuration files,
802 static Realtime, and dynamic Realtime. It also provides a caching service.
803 Users can configure a hierarchy of data storage layers for specific modules
806 * All future modules which utilize Sorcery for object persistence must have a
807 column named "id" within their schema when using the Sorcery realtime module.
808 This column must be able to contain a string of up to 128 characters in length.
810 Security Events Framework
812 * Security Event timestamps now use ISO 8601 formatted date/time instead of
813 the "seconds-microseconds" format that it was using previously.
817 * The Stasis message bus is a publish/subscribe message bus internal to
818 Asterisk. Many services in Asterisk are built on the Stasis message bus,
819 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
820 Stasis can be configured in stasis.conf. Note that these parameters operate
821 at a very low level in Asterisk, and generally will not require changes.
825 * When a channel driver is configured to enable jiterbuffers, they are now
826 applied unconditionally when a channel joins a bridge. If a jitterbuffer
827 is already set for that channel when it enters, such as by the JITTERBUFFER
828 function, then the existing jitterbuffer will be used and the one set by
829 the channel driver will not be applied.
833 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
834 dialplan applications provided by the app_agent_pool module. Agents are
835 connected with callers using the new AgentRequest dialplan application.
836 The Agents:<agent-id> device state is available to monitor the status of an
837 agent. See agents.conf.sample for valid configuration options.
839 * The updatecdr option has been removed. Altering the names of channels on a
840 CDR is not supported - the name of the channel is the name of the channel,
841 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
842 has also been removed, for the same reason.
844 * The endcall and enddtmf configuration options are removed. Use the
845 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
846 channel before calling AgentLogin.
850 * chan_bridge has been removed. Its functionality has been incorporated
851 directly into the ConfBridge application itself.
855 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
856 of the specified span and its B-channels. Note that this command should
857 only be used if you understand the risks it entails.
859 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
860 A range of channels can be specified to be destroyed. Note that this command
861 should only be used if you understand the risks it entails.
863 * Added the CLI command 'dahdi create channels'. A range of channels can be
864 specified to be created, or the keyword 'new' can be used to add channels
869 * IPv6 support has been added. We are now able to bind to and
870 communicate using IPv6 addresses.
874 * The /b option has been removed.
876 * chan_local moved into the system core and is no longer a loadable module.
880 * Added general support for busy detection.
882 * Added ECAM command support for Sony Ericsson phones.
886 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
887 SIP stack. A collection of resource modules provides the bulk of the SIP
888 functionality. For more information on the new SIP channel driver, see
889 https://wiki.asterisk.org/wiki/x/JYGLAQ
893 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
894 using the 'supportpath' setting, either on a global basis or on a peer basis.
895 This setting enables Asterisk to route outgoing out-of-dialog requests via a
896 set of proxies by using a pre-loaded route-set defined by the Path headers in
897 the REGISTER request. See Realtime updates for more configuration information.
899 * The SIP_CODEC family of variables may now specify more than one codec. Each
900 codec must be separated by a comma. The first codec specified is the
901 preferred codec for the offer. This allows a dialplan writer to specify both
902 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
904 * The 'callevents' parameter has been removed. Hold AMI events are now raised
905 in the core, and can be filtered out using the 'eventfilter' parameter
908 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
909 codecs configured for a peer instead of the requested codec.
911 * The option "register_retry_403" has been added to chan_sip to work around
912 servers that are known to erroneously send 403 in response to valid
913 REGISTER requests and allows Asterisk to continue attepmting to connect.
917 * Added the 'immeddialkey' parameter. If set, when the user presses the
918 configured key the already entered number will be immediately dialed. This
919 is useful when the dialplan allows for variable length pattern matching.
920 Valid options are '*' and '#'.
922 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
923 milliseconds) before a call forward is considered to not be answered.
925 * The 'serviceurl' parameter allows Service URLs to be attached to line
934 * The password option has been disabled, as the AgentLogin application no
935 longer provides authentication.
939 * Due to changes in the Asterisk core, this function is no longer needed to
940 preserve a MixMonitor on a channel during transfer operations and dialplan
941 execution. It is effectively obsolete.
945 * The 'amaflags' and 'accountcode' attributes for the CDR function are
946 deprecated. Use the CHANNEL function instead to access these attributes.
948 * The 'l' option has been removed. When reading a CDR attribute, the most
949 recent record is always used. When writing a CDR attribute, all non-finalized
952 * The 'r' option has been removed, for the same reason as the 'l' option.
954 * The 's' option has been removed, as LOCKED semantics no longer exist in the
959 * A new function CDR_PROP has been added. This function lets you set properties
960 on a channel's active CDRs. This function is write-only. Properties accept
961 boolean values to set/clear them on the channel's CDRs. Valid properties
963 - 'party_a' - make this channel the preferred Party A in any CDR between two
964 channels. If two channels have this property set, the creation time of the
965 channel is used to determine who is Party A. Note that dialed channels are
966 never Party A in a CDR.
967 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
968 application when set to True, and analogous to the 'e' option in ResetCDR
973 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
974 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
975 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
978 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
979 string, i.e., [[context],extension],priority. If set on a channel, if a
980 channel leaves a bridge but is not hung up it will resume dialplan execution
985 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
986 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
987 The value of this setting is ignored when disabled is used for the argument.
991 * A new function provided by chan_pjsip, this function can be used in
992 conjunction with the Dial application to construct a dial string that will
993 dial all contacts on an Address of Record associated with a chan_pjsip
998 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
999 outbound channel prior to dialing.
1003 * Redirecting reasons can now be set to arbitrary strings. This means
1004 that the REDIRECTING dialplan function can be used to set the redirecting
1005 reason to any string. It also allows for custom strings to be read as the
1006 redirecting reason from SIP Diversion headers.
1010 * The SPEECH_ENGINE function now supports read operations. When read from, it
1011 will return the current value of the requested attribute.
1017 res_agi (Asterisk Gateway Interface)
1019 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1021 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1024 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1025 will start the playback of the audio at the position specified. It will
1026 also return the final position of the file in 'endpos'.
1028 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1029 channel variable if the user stopped the file playback or if a remote
1030 entity stopped the playback. If neither stopped the playback, it will
1031 indicate the overall success/failure of the playback. If stopped early,
1032 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1035 * The SAY ALPHA command now accepts an additional parameter to control
1036 whether it specifies the case of uppercase, lowercase, or all letters to
1037 provide functionality similar to SayAlphaCase.
1039 res_ari (Asterisk RESTful Interface) (and others)
1041 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1042 control telephony primitives in Asterisk by remote client. This includes
1043 channels, bridges, endpoints, media, and other fundamental concepts. Users
1044 of ARI can develop their own communications applications, controlling
1045 multiple channels using an HTTP RESTful interface and receiving JSON events
1046 about the objects via a WebSocket connection. ARI can be configured in
1047 Asterisk via ari.conf. For more information on ARI, see
1048 https://wiki.asterisk.org/wiki/x/0YCLAQ
1052 * Parking has been extracted from the Asterisk core as a loadable module,
1053 res_parking. Configuration for parking is now provided by res_parking.conf.
1054 Configuration through features.conf is no longer supported.
1056 * res_parking uses the configuration framework. If an invalid configuration is
1057 supplied, res_parking will fail to load or fail to reload. Previously,
1058 invalid configurations would generally be accepted, with certain errors
1059 resulting in individually disabled parking lots.
1061 * Parked calls are now placed in bridges. While this is largely an
1062 architectural change, it does have implications on how channels in a parking
1063 lot are viewed. For example, commands that display channels in bridges will
1064 now also display the channels in a parking lot.
1066 * The order of arguments for the new parking applications have been modified.
1067 Timeout and return context/exten/priority are now implemented as options,
1068 while the name of the parking lot is now the first parameter. See the
1069 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1070 in-depth information as well as syntax.
1072 * Extensions are by default no longer automatically created in the dialplan to
1073 park calls or pickup parked calls. Generation of dialplan extensions can be
1074 enabled using the 'parkext' configuration option.
1076 * ADSI functionality for parking is no longer supported. The 'adsipark'
1077 configuration option has been removed as a result.
1079 * The PARKINGSLOT channel variable has been deprecated in favor of
1080 PARKING_SPACE to match the naming scheme of the new system.
1082 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1083 channel even when the configuration option 'comebactoorigin' is enabled.
1085 * A new CLI command 'parking show' has been added. This allows a user to
1086 inspect the parking lots that are currently in use.
1087 'parking show <parkinglot>' will also show the parked calls in a specific
1090 * The CLI command 'parkedcalls' is now deprecated in favor of
1091 'parking show <parkinglot>'.
1093 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1094 can be used to get a list of parked calls for a specific parking lot.
1096 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1097 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1098 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1099 longer a required argument.
1101 * The ParkAndAnnounce application is now provided through res_parking instead
1102 of through the separate app_parkandannounce module.
1104 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1105 by default. Instead, it will follow the timeout rules of the parking lot. The
1106 old behavior can be reproduced by using the 'c' option.
1108 * Dynamic parking lots will now fail to be created under the following
1110 - if the parking lot specified by PARKINGDYNAMIC does not exist
1111 - if they require exclusive park and parkedcall extensions which overlap
1112 with existing parking lots.
1114 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1115 currently contain no calls. Dynamic parking lots containing parked calls
1116 will persist through the reloads without alteration.
1118 * If 'parkext_exclusive' is set for a parking lot and that extension is
1119 already in use when that parking lot tries to register it, this is now
1120 considered a parking system configuration error. Configurations which do
1121 this will be rejected.
1123 * Added channel variable PARKER_FLAT. This contains the name of the extension
1124 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1125 comebacktoorigin is disabled, but the dialplan or an external control
1126 mechanism wants to use the extension in the park-dial context that was
1127 generated to re-dial the parker on timeout.
1129 res_pjsip (and many others)
1131 * A large number of resource modules make up the SIP stack based on pjsip.
1132 The chan_pjsip channel driver users these resource modules to provide
1133 various SIP functionality in Asterisk. The majority of configuration for
1134 these modules is performed in pjsip.conf. Other modules may use their
1135 own configuration files.
1139 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1140 them, an Asterisk-specific version of PJSIP needs to be installed.
1141 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1143 res_statsd/res_chan_stats
1145 * A new resource module, res_statsd, has been added, which acts as a statsd
1146 client. This module allows Asterisk to publish statistics to a statsd
1147 server. In conjunction with res_chan_stats, it will publish statistics about
1148 channels to the statsd server. It can be configured via res_statsd.conf.
1152 * Device state for XMPP buddies is now available using the following format:
1153 XMPP/<client name>/<buddy address>
1154 If any resource is available the device state is considered to be not in use.
1155 If no resources exist or all are unavailable the device state is considered
1162 Realtime/Database Scripts
1164 * Asterisk previously included example db schemas in the contrib/realtime/
1165 directory of the source tree. This has been replaced by a set of database
1166 migrations using the Alembic framework. This allows you to use alembic to
1167 initialize the database for you. It will also serve as a database migration
1168 tool when upgrading Asterisk in the future.
1170 See contrib/ast-db-manage/README.md for more details.
1174 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1175 This python script will convert an existing sip.conf file to a
1176 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1177 is meant to be an aid in converting an existing chan_sip configuration to
1178 a chan_pjsip configuration, but it is expected that configuration beyond
1179 what the script provides will be needed.
1182 ------------------------------------------------------------------------------
1183 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1184 ------------------------------------------------------------------------------
1188 * The Asterisk build system will now build and install a shared library
1189 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1190 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1191 that Asterisk can ensure that these functions do *not* get called by any
1192 modules that are loaded into Asterisk, since they should only be called once
1193 in any single process. If desired, this feature can be disabled by supplying
1194 the "--disable-asteriskssl" option to the configure script.
1196 * A new make target, 'full', has been added to the Makefile. This performs
1197 the same compilation actions as make all, but will also scan the entirety of
1198 each source file for documentation. This option is needed to generate AMI
1199 event documentation. Note that your system must have Python in order for
1200 this make target to succeed.
1202 * The optimization portion of the build system has been reworked to avoid
1203 broken builds on certain architectures. All architecture-specific
1204 optimization has been removed in favor of using -march=native to allow gcc
1205 to detect the environment in which it is running when possible. This can
1206 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1208 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1209 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1211 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1212 previously parsed the header file to obtain the version of Asterisk, you
1213 will now have to go through Asterisk to get the version information.
1221 * Added 'F()' option. Similar to the dial option, this can be supplied with
1222 arguments indicating where the callee should go after the caller is hung up,
1223 or without options specified, the priority after the Queue will be used.
1228 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1229 all non-admin participants on a conference. The confbridge configuration
1230 file also allows for the default sounds played to all conference users when
1231 this occurs to be overriden using sound_participants_unmuted and
1232 sound_participants_muted.
1234 * Added menu action participant_count. This will playback the number of
1235 current participants in a conference.
1237 * Added announcement configuration option to user profile. If set the sound
1238 file will be played to the user, and only the user, upon joining the
1241 * Added record_file_append option that defaults to "yes", but if set to no
1242 will create a new file between each start/stop recording.
1247 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1248 channels respectively before the callee channels are called.
1253 * Added support for IPv6.
1255 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1256 external process will cause the current playlist to be cleared, including
1257 stopping any audio file that is currently playing. This is useful when you
1258 want to interrupt audio playback only when specific DTMF is entered by the
1264 * A new option, 'I' has been added to app_followme. By setting this option,
1265 Asterisk will not update the caller with connected line changes when they
1266 occur. This is similar to app_dial and app_queue.
1268 * The 'N' option is now ignored if the call is already answered.
1270 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1271 and caller channels respectively before the callee channels are called.
1273 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1279 * MixMonitor hooks now have IDs associated with them which can be used to
1280 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1281 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1282 now accepts that ID as an argument.
1284 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1285 indicated mailboxes.
1290 * The connect action in app_mysql now allows you to specify a port number to
1291 connect to. This is useful if you run a MySQL server on a non-standard
1297 * Increased the default number of allowed destinations from 5 to 12.
1302 * The app_page application now no longer depends on DAHDI or app_meetme. It
1303 has been re-architected to use app_confbridge internally.
1308 * Added queue options autopausebusy and autopauseunavail for automatically
1309 pausing a queue member when their device reports busy or congestion.
1311 * The 'ignorebusy' option for queue members has been deprecated in favor of
1312 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1313 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1314 per interface basis. Individual ringinuse values can now be set in
1315 queues.conf via an argument to member definitions. Lastly, the queue
1316 'ringinuse' setting now only determines defaults for the per member
1317 'ringinuse' setting and does not override per member settings like it does
1318 in earlier versions.
1320 * Added 'F()' option. Similar to the dial option, this can be supplied with
1321 arguments indicating where the callee should go after the caller is hung up,
1322 or without options specified, the priority after the Queue will be used.
1324 * Added new option log_member_name_as_agent, which will cause the membername to
1325 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1326 state_interface has been set.
1328 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1330 * App_queue will now play periodic announcements for the caller that
1331 holds the first position in the queue while waiting for answer.
1335 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1336 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1337 changed arguments to SayUnixTime so that every option is truly optional even
1338 when using multiple options (so that j option could be used without having to
1339 manually specify timezone and format) There are other benefits, e.g., format
1340 can now be used without specifying time zone as well.
1345 * Addition of the VM_INFO function - see Function changes.
1347 * The imapserver, imapport, and imapflags configuration options can now be
1348 overriden on a user by user basis.
1350 * When voicemail plays a message's envelope with saycid set to yes, when
1351 reaching the caller id field it will play a recording of a file with the same
1352 base name as the sender's callerid if there is a similarly named file in
1353 <astspooldir>/recordings/callerids/
1355 * Voicemails now contains a unique message identifier "msg_id", which is stored
1356 in the message envelope with the sound files. IMAP backends will now store
1357 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1358 backends will store the message identifier in a "msg_id" column. See
1359 UPGRADE.txt for more information.
1361 * Added VoiceMailPlayMsg application. This application will play a single
1362 voicemail message from a mailbox. The result of the application, SUCCESS or
1363 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1368 * Hangup handlers can be attached to channels using the CHANNEL() function.
1369 Hangup handlers will run when the channel is hung up similar to the h
1370 extension. The hangup_handler_push option will push a GoSub compatible
1371 location in the dialplan onto the channel's hangup handler stack. The
1372 hangup_handler_pop option will remove the last added location, and optionally
1373 replace it with a new GoSub compatible location. The hangup_handler_wipe
1374 option will remove all locations on the stack, and optionally add a new
1377 * The expression parser now recognizes the ABS() absolute value function,
1378 which will convert negative floating point values to positive values.
1380 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1381 control of faxdetect.
1383 * Addition of the VM_INFO function that can be used to retrieve voicemail
1384 user information, such as the email address and full name.
1385 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1388 * The REDIRECTING function now supports the redirecting original party id
1391 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1392 lets you set some of the configuration options from the [general] section
1393 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1394 the key sequence used to activate built-in features, such as blindxfer,
1395 and automon. See the built-in documentation for details.
1397 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1398 instead of simply the uri. This is the format that MessageSend() can use
1399 in the from parameter for outgoing SIP messages.
1401 * Added the PRESENCE_STATE function. This allows retrieving presence state
1402 information from any presence state provider. It also allows setting
1403 presence state information from a CustomPresence presence state provider.
1404 See AMI/CLI changes for related commands.
1406 * Added the AMI_CLIENT function to make manager account attributes available
1407 to the dialplan. It currently supports returning the current number of
1408 active sessions for a given account.
1410 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1411 and the REDIRECTING functions.
1419 * Added a manager event "LocalBridge" for local channel call bridges between
1420 the two pseudo-channels created.
1425 * Added dialtone_detect option for analog ports to disconnect incoming
1426 calls when dialtone is detected.
1428 * Added option colp_send to send ISDN connected line information. Allowed
1429 settings are block, to not send any connected line information; connect, to
1430 send connected line information on initial connect; and update, to send
1431 information on any update during a call. Default is update.
1433 * Add options namedcallgroup and namedpickupgroup to support installations
1434 where a higher number of groups (>64) is required.
1436 * Added support to use private party ID information with PRI calls.
1441 * A new channel driver named chan_motif has been added which provides support for
1442 Google Talk and Jingle in a single channel driver. This new channel driver includes
1443 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1444 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1445 specification, current Google Jingle specification, and the original Google Talk
1451 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1452 globally and overriden on a peer by peer basis.
1454 * Direct media functionality has been added. Options in config are:
1455 directmedia (directrtp) and directrtpsetup (earlydirect)
1457 * ChannelUpdate events now contain a CallRef header.
1462 * Asterisk will no longer substitute CID number for CID name in the display
1463 name field if CID number exists without a CID name. This change improves
1464 compatibility with certain device features such as Avaya IP500's directory
1467 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1468 created using that setting to not be removed during SIP reload.
1470 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1471 request with a "Record:" header, this will turn the requested feature on/off.
1472 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1473 dynamic features must be enabled and configured properly on the requesting
1474 channel for this to function properly.
1476 * Add support to realtime for the 'callbackextension' option.
1478 * When multiple peers exist with the same address, but differing
1479 callbackextension options, incoming requests that are matched by address
1480 will be matched to the peer with the matching callbackextension if it is
1483 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1484 which set the force_rport and comedia options automatically if Asterisk
1485 detects that an incoming SIP request crossed a NAT after being sent by
1486 the remote endpoint.
1488 * The default global nat setting in sip.conf has been changed from force_rport
1489 to auto_force_rport.
1491 * NAT settings are now a combinable list of options. The equivalent of the
1492 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1494 * Adds an option send_diversion which can be disabled to prevent
1495 diversion headers from automatically being added to INVITE requests.
1497 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1498 be sent to the remote host at a given interval to keep the NAT mapping open.
1499 This can be enabled using the keepalive configuration option.
1501 * Add option 'tonezone' to specify country code for indications. This option
1502 can be set both globally and overridden for specific peers.
1504 * The SIP Security Events Framework now supports IPv6.
1506 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1507 between multiple user agents. When set, for directmedia reinvites,
1508 Asterisk will not send an immediate reinvite on an incoming call leg. This
1509 option is useful when peered with another SIP user agent that is known to
1510 send immediate direct media reinvites upon call establishment.
1512 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1515 * Add options subminexpiry and submaxexpiry to set limits of subscription
1516 timer independently from registration timer settings. The setting of the
1517 registration timer limits still is done by options minexpiry, maxexpiry
1518 and defaultexpiry. For backwards compatibility the setting of minexpiry
1519 and maxexpiry also is used to configure the subscription timer limits if
1520 subminexpiry and submaxexpiry are not set in sip.conf.
1522 * Set registration timer limits to default values when reloading sip
1523 configuration and values are not set by configuration.
1525 * Add options namedcallgroup and namedpickupgroup to support installations
1526 where a higher number of groups (>64) is required.
1528 * When a MESSAGE request is received, the address the request was received from
1529 is now saved in the SIP_RECVADDR variable.
1531 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1532 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1533 the ANI2/OLI information is set on the channel, which can be retrieved using
1534 the CALLERID function.
1536 * Peers can now be configured to support negotiation of ICE candidates using
1537 the setting icesupport. See res_rtp_asterisk changes for more information.
1539 * Added support for format attribute negotiation. See the Codecs changes for
1542 * Extra headers specified with SIPAddHeader are sent with the REFER message
1543 when using Transfer application. See refer_addheaders in sip.conf.sample.
1545 * Added support to use private party ID information with calls.
1547 * Adds an option discard_remote_hold_retrieval that when set stops telling
1548 the peer to start music on hold.
1553 * Added skinny version 17 protocol support.
1557 --------------------
1558 * Added ability to use multiple lines for a single phone. This allows multiple
1559 calls to occur on a single phone, using callwaiting and switching between calls.
1561 * Added option 'sharpdial' allowing end dialing by pressing # key
1563 * Added option 'interdigit_timer' to control phone dial timeout
1565 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1567 * Added global 'debug' option, that enables debug in channel driver
1569 * Added ability to translate on-screen menu in multiple languages. Tested on
1570 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1571 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1574 * In addition to English added French and Russian languages for on-screen menus
1576 * Reworked dialing number input: added dialing by timeout, immediate dial on
1577 on dialplan compare, phone number length now not limited by screen size
1579 * Added ability to pickup a call using features.conf defined value and
1585 * Add options namedcallgroup and namedpickupgroup to support installations
1586 where a higher number of groups (>64) is required.
1588 * Added support to use private party ID information with calls.
1593 * The minimum DTMF duration can now be configured in asterisk.conf
1594 as "mindtmfduration". The default value is (as before) set to 80 ms.
1595 (previously it was only available in source code)
1597 * Named ACLs can now be specified in acl.conf and used in configurations that
1598 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1599 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1600 working ACL. In addition, some CLI commands have been added to provide
1601 show information and allow for module reloading - see CLI Changes.
1603 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1604 items (separated by commas), and items in the rule can be negated by prefixing
1605 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1606 longer necessray to control the order that the 'permit' and 'deny' columns are
1607 returned from queries.
1609 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1610 be used within the dynamic weight attribute when specifying a mapping.
1612 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1613 header, instead of putting the user defined event name there. When enabled
1614 the UserDefType header is added for user defined events. This feature is
1615 enabled with the setting show_user_defined.
1617 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1618 line purposes use the following variables instead of their macro equivalents:
1619 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1620 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1621 cc_callback_macro in channel configurations.
1623 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1626 * Call files now support the "early_media" option to connect with an outgoing
1627 extension when early media is received.
1629 * Added support to use private party ID information with calls.
1634 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1635 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1636 AGI application would exit immediately after a channel hangup is detected.
1638 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1639 are resolved and each address is attempted in turn until one succeeds or
1643 AMI (Asterisk Manager Interface)
1645 * The originate action now has an option "EarlyMedia" that enables the
1646 call to bridge when we get early media in the call. Previously,
1647 early media was disregarded always when originating calls using AMI.
1649 * Added setvar= option to manager accounts (much like sip.conf)
1651 * Originate now generates an error response if the extension given is not found
1654 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1655 them if the i(variable) option is used. StopMixMonitor will accept
1656 MixMonitorID as an option to close specific MixMonitors.
1658 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1659 updated to include information about peers configured with
1660 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1661 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1662 returned if auto_force_rport is not enabled.
1664 * Added SIPpeerstatus manager command which will generate PeerStatus events
1665 similar to the existing PeerStatus events found in chan_sip on demand.
1667 * Hangup now can take a regular expression as the Channel option. If you want
1668 to hangup multiple channels, use /regex/ as the Channel option. Existing
1669 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1670 the manager will send you a list of channels back that were hung up.
1672 * Support for IPv6 addresses has been added.
1674 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1675 documentation is only generated when Asterisk is compiled using 'make full'.
1676 See the CLI section for commands to display AMI event information.
1678 * The AMI Hangup event now includes the AccountCode header so you can easily
1679 correlate with AMI Newchannel events.
1681 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1682 the StateInterface of the queue member.
1684 * Added AMI event SessionTimeout in the Call category that is issued when a
1685 call is terminated due to either RTP stream inactivity or SIP session timer
1688 * CEL events can now contain a user defined header UserDefType. See core
1689 changes for more information.
1691 * OOH323 ChannelUpdate events now contain a CallRef header.
1693 * Added PresenceState command. This command will report the presence state for
1694 the given presence provider.
1696 * Added Parkinglots command. This will list all parking lots as a series of
1697 AMI Parkinglot events.
1699 * Added MessageSend command. This behaves in the same manner as the
1700 MessageSend application, and is a technolgoy agnostic mechanism to send out
1701 of call text messages.
1703 * Added "message" class authorization. This grants an account permission to
1704 send out of call messages. Write-only.
1709 * The "dialplan add include" command has been modified to create context a context
1710 if one does not already exist. For instance, "dialplan add include foo into bar"
1711 will create context "bar" if it does not already exist.
1713 * A "dialplan remove context" command has been added to remove a context from
1716 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1717 filenames of all running mixmonitors on a channel.
1719 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1720 numeric instead of 0, 1, or 2.
1722 * "stun show status" will show a table describing how the STUN client is
1725 * "acl show [named acl]" will show information regarding a Named ACL. The
1726 acl module can be reloaded with "reload acl".
1728 * Added CLI command to display AMI event information - "manager show events",
1729 which shows a list of all known and documented AMI events, and "manager show
1730 event [event name]", which shows detail information about a specific AMI
1733 * The result of the CLI command "queue show" now includes the state interface
1734 information of the queue member.
1736 * The command "core set verbose" will now set a separate level of logging for
1737 each remote console without affecting any other console.
1739 * Added command "cdr show pgsql status" to check connection status
1741 * "sip show channel" will now display the complete route set.
1743 * Added "presencestate list" command. This command will list all custom
1744 presence states that have been set by using the PRESENCE_STATE dialplan
1747 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1748 command. This changes a custom presence to a new state.
1753 * Codec lists may now be modified by the '!' character, to allow succinct
1754 specification of a list of codecs allowed and disallowed, without the
1755 requirement to use two different keywords. For example, to specify all
1756 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1758 * Add support for parsing SDP attributes, generating SDP attributes, and
1759 passing it through. This support includes codecs such as H.263, H.264, SILK,
1760 and CELT. You are able to set up a call and have attribute information pass.
1761 This should help considerably with video calls.
1763 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1764 just like the GSM codec.
1768 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1769 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1773 * Asterisk version and build information is now logged at the beginning of a
1776 * Threads belonging to a particular call are now linked with callids which get
1777 added to any log messages produced by those threads. Log messages can now be
1778 easily identified as involved with a certain call by looking at their call id.
1779 Call ids may also be attached to log messages for just about any case where
1780 it can be determined to be related to a particular call.
1782 * Each logging destination and console now have an independent notion of the
1783 current verbosity level. Logger.conf now allows an optional argument to
1784 the 'verbose' specifier, indicating the level of verbosity sent to that
1785 particular logging destination. Additionally, remote consoles now each
1786 have their own verbosity level. The command 'core set verbose' will now set
1787 a separate level for each remote console without affecting any other
1793 * Added 'announcement' option which will play at the start of MOH and between
1794 songs in modes of MOH that can detect transitions between songs (eg.
1800 * New per parking lot options: comebackcontext and comebackdialtime. See
1801 configs/features.conf.sample for more details.
1803 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1806 * Channel variable PARKEDCALL is now set with the name of the parking lot
1807 when a timeout occurs.
1813 CDR Postgresql Driver
1815 * Added command "cdr show pgsql status" to check connection status
1818 CDR Adaptive ODBC Driver
1820 * Added schema option for databases that support specifying a schema.
1828 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1829 CALENDAR_WRITE has completed successfully.
1834 * A new option, 'probation' has been added to rtp.conf
1835 RTP in strictrtp mode can now require more than 1 packet to exit learning
1836 mode with a new source (and by default requires 4). The probation option
1837 allows the user to change the required number of packets in sequence to any
1838 desired value. Use a value of 1 to essentially restore the old behavior.
1839 Also, with strictrtp on, Asterisk will now drop all packets until learning
1840 mode has successfully exited. These changes are based on how pjmedia handles
1841 media sources and source changes.
1843 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1844 enabled or disabled using the icesupport setting. A variety of other
1845 settings have been introduced to configure STUN/TURN connections.
1850 * A new module, res_corosync, has been introduced. This module uses the
1851 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1852 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1853 Device State (presence) information. This module is very similar to, and
1854 is a replacement for the res_ais module that was in previous releases of
1860 * This module adds a cleaned up, drop-in replacement for res_jabber called
1861 res_xmpp. This provides the same externally facing functionality but is
1862 implemented differently internally. res_jabber has been deprecated in favor
1863 of res_xmpp; please see the UPGRADE.txt file for more information.
1868 * The safe_asterisk script has been updated to allow several of its parameters
1869 to be set from environment variables. This also enables a custom run
1870 directory of Asterisk to be specified, instead of defaulting to /tmp.
1872 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1873 its value to determine the directory to assume is the top-level directory of
1874 the source tree. If the variable is not set, it defaults to the current
1875 behavior and uses the current working directory.
1877 ------------------------------------------------------------------------------
1878 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1879 ------------------------------------------------------------------------------
1883 * Asterisk now has protocol independent support for processing text messages
1884 outside of a call. Messages are routed through the Asterisk dialplan.
1885 SIP MESSAGE and XMPP are currently supported. There are options in
1886 jabber.conf and sip.conf to allow enabling these features.
1887 -> jabber.conf: see the "sendtodialplan" and "context" options.
1888 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1889 and "outofcall_message_context" options.
1890 The MESSAGE() dialplan function and MessageSend() application have been
1891 added to go along with this functionality. More detailed usage information
1892 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1893 * If real-time text support (T.140) is negotiated, it will be preferred for
1894 sending text via the SendText application. For example, via SIP, messages
1895 that were once sent via the SIP MESSAGE request would be sent via RTP if
1896 T.140 text is negotiated for a call.
1900 * parkedmusicclass can now be set for non-default parking lots.
1902 Asterisk Manager Interface
1903 --------------------------
1904 * PeerStatus now includes Address and Port.
1905 * Added Hold events for when the remote party puts the call on and off hold
1906 for chan_dahdi ISDN channels.
1907 * Added new action MeetmeListRooms to list active conferences (shows same
1908 data as "meetme list" at the CLI).
1909 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1910 Description field that is set by 'description' in the channel configuration
1912 * Added Uniqueid header to UserEvent.
1913 * Added new action FilterAdd to control event filters for the current session.
1914 This requires the system permission and uses the same filter syntax as
1915 filters that can be defined in manager.conf
1916 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1917 versions had some instances of the event converted, but others were left
1918 as-is. All Unlink events should now be converted to Bridge events. The AMI
1919 protocol version number was incremented to 1.2 as a result of this change.
1921 Asterisk HTTP Server
1922 --------------------------
1923 * The HTTP Server can bind to IPv6 addresses.
1926 --------------------------
1927 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1928 with busydetect. usage example: busypattern=200,200,200,600
1931 --------------------------
1932 * New 'gtalk show settings' command showing the current settings loaded from
1934 * The 'logger reload' command now supports an optional argument, specifying an
1935 alternate configuration file to use.
1936 * 'dialplan add extension' command will now automatically create a context if
1937 the specified context does not exist with a message indicated it did so.
1938 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1939 Description field which can be populated with 'description' in the channel
1940 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1943 --------------------------
1944 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1945 thus allowing records which do NOT match the specified filter.
1946 * Added ability to log CONGESTION calls to CDR
1949 --------------------------
1950 * Ability to define custom SILK formats in codecs.conf.
1951 * Addition of speex32 audio format with translation.
1952 * CELT codec pass-through support and ability to define
1953 custom CELT formats in codecs.conf.
1954 * Ability to read raw signed linear files with sample rates
1955 ranging from 8khz - 192khz. The new file extensions introduced
1956 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1957 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1958 Skinny, H.323, etc) can still only support the following codecs:
1959 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1960 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1961 Video: h261, h263, h263p, h264, mpeg4
1966 --------------------------
1967 * New highly optimized and customizable ConfBridge application capable of
1968 mixing audio at sample rates ranging from 8khz-96khz.
1969 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1970 and bridge profiles on a channel.
1971 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1972 about a conference such as locked status and number of parties, admins,
1974 * Addition of video_mode option in confbridge.conf for adding video support
1975 into a bridge profile.
1976 * Addition of the follow_talker video_mode in confbridge.conf. This video
1977 mode dynamically switches the video feed to always display the loudest talker
1978 supplying video in the conference.
1982 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1983 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1984 variables from asterisk.conf.
1988 * Addition of the JITTERBUFFER dialplan function. This function allows
1989 for jitterbuffering to occur on the read side of a channel. By using
1990 this function conference applications such as ConfBridge and MeetMe can
1991 have the rx streams jitterbuffered before conference mixing occurs.
1992 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
1994 * Added STRREPLACE function. This function let's the user search a variable
1995 for a given string to replace with another string as many times as the
1996 user specifies or just throughout the whole string.
1997 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
1998 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
1999 * Added extensions to chan_ooh323 in function CHANNEL()
2001 libpri channel driver (chan_dahdi) DAHDI changes
2002 --------------------------
2003 * Added moh_signaling option to specify what to do when the channel's bridged
2004 peer puts the ISDN channel on hold.
2005 * Added display_send and display_receive options to control how the display ie
2006 is handled. To send display text from the dialplan use the SendText()
2007 application when the option is enabled.
2008 * Added mcid_send option to allow sending a MCID request on a span.
2011 --------------------------
2012 * Added setvar option to calendar.conf to allow setting channel variables on
2013 notification channels.
2014 * Added "calendar show types" CLI command to list registered calendar
2018 --------------------------
2019 * Added two new options, r and t with file name arguments to record
2020 single direction (unmixed) audio recording separate from the bidirectional
2021 (mixed) recording. The mixed file name argument is optional now as long
2022 as at least one recording option is used.
2025 --------------------------
2026 * Added a new option, l, which will disable local call optimization for
2027 channels involved with the FollowMe thread. Use this option to improve
2028 compatability for a FollowMe call with certain dialplan apps, options, and
2032 --------------------------
2033 * Added option "k" that will automatically close the conference when there's
2034 only one person left when a user exits the conference.
2037 --------------------------
2038 * cel_pgsql now supports the 'extra' column for data added using the
2039 CELGenUserEvent() application.
2042 --------------------------
2043 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2044 in the sample extensions.lua file for syntax details.
2045 * Applications that perform jumps in the dialplan such as Goto will now
2046 execute properly. When pbx_lua detects that the context, extension, or
2047 priority we are executing on has changed it will immediately return control
2048 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2049 the priority after the currently executing priority.
2050 * An autoservice is now started by default for pbx_lua channels. It can be
2051 stopped and restarted using the autoservice_stop() and autoservice_start()
2055 --------------------------
2056 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2057 into a FAXStatus event with an 'Operation' header that will be either
2058 'send', 'receive', and 'gateway'.
2059 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2060 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2061 feature will handle converting a fax call between an audio T.30 fax terminal
2062 and an IFP T.38 fax terminal.
2066 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2067 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2068 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2072 * Added general option negative_penalty_invalid default off. when set
2073 members are seen as invalid/logged out when there penalty is negative.
2074 for realtime members when set remove from queue will set penalty to -1.
2075 * Added queue option autopausedelay when autopause is enabled it will be
2076 delayed for this number of seconds since last successful call if there
2077 was no prior call the agent will be autopaused immediately.
2078 * Added member option ignorebusy this when set and ringinuse is not
2079 will allow per member control of multiple calls as ringinuse does for
2084 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2086 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2087 one participant left (much like a normal call bridge)
2088 * Added extra argument to Originate to set timeout.
2092 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2093 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2094 utility in the UTILS section of menuselect. If an existing astdb is found and no
2095 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2096 convert an existing astdb to the SQLite3 version automatically at runtime.
2100 * Modules marked as deprecated are no longer marked as building by default. Enabling
2101 these modules is still available via menuselect.
2105 * authdebug is now disabled by default. To enable this functionaility again
2106 set authdebug = yes in iax.conf.
2110 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2111 releases it was disabled.
2115 * The PBX core previously made a call with a non-existing extension test for
2116 extension s@default and jump there if the extension existed.
2117 This was a bad default behaviour and violated the principle of least surprise.
2118 It has therefore been changed in this release. It may affect some
2119 applications and configurations that rely on this behaviour. Most channel
2120 drivers have avoided this for many releases by testing whether the extension
2121 called exists before starting the PBX and generating a local error.
2122 This behaviour still exists and works as before.
2124 Extension "s" is used when no extension is given in a channel driver,
2125 like immediate answer in DAHDI or calling to a domain with no user part
2128 ------------------------------------------------------------------------------
2129 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2130 ------------------------------------------------------------------------------
2134 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2135 now defaults to force_rport. It is very important that phones requiring nat=no be
2136 specifically set as such instead of relying on the default setting. If at all
2137 possible, all devices should have nat settings configured in the general section as
2138 opposed to configuring nat per-device.
2139 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2140 codecs sent in response to an INVITE to the single most preferred codec.
2141 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2142 to be used for the outgoing call. It must be one of the codecs configured
2144 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2145 to be used for holding a private key. If tlsprivatekey is not specified,
2146 tlscertfile is searched for both public and private key.
2147 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2148 outbound client connections to be specified.
2149 * The sendrpid parameter has been expanded to include the options
2150 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2151 header to be sent (equivalent to setting sendrpid=yes) and setting
2152 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2153 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2154 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2155 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2156 will accept the SDP even if the SDP version number is not properly incremented,
2157 but will generate a warning in the log indicating that the SIP peer that sent
2158 the SDP should have the 'ignoresdpversion' option set.
2159 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2160 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2161 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2162 remote side requests it and disables symmetric RTP support. Setting it to
2163 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2164 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2165 and enables symmetric RTP support.
2166 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2167 response. This permits the master channel to know how each channel dialled
2168 in a multi-channel setup resolved in an individual way. This carries a
2169 performance penalty and can be disabled in sip.conf using the
2170 'storesipcause' option.
2171 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2172 configuration for the externip and externhost options when tcp or tls is used.
2173 * Added support for message body (stored in content variable) to SIP NOTIFY message
2174 accessible via AMI and CLI.
2175 * Added 'media_address' configuration option which can be used to explicitly specify
2176 the IP address to use in the SDP for media (audio, video, and text) streams.
2177 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2178 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2180 * Added 'use_q850_reason' configuration option for generating and parsing
2181 if available Reason: Q.850;cause=<cause code> header. It is implemented
2182 in some gateways for better passing PRI/SS7 cause codes via SIP.
2183 * When dialing SIP peers, a new component may be added to the end of the dialstring
2184 to indicate that a specific remote IP address or host should be used when dialing
2185 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2186 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2187 ability to selectively force bridged channels to also be encrypted is also
2188 implemented. Branching in the dialplan can be done based on whether or not
2189 a channel has secure media and/or signaling.
2190 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2192 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2193 Charge messages to snom phones.
2194 * Added support for G.719 media streams.
2195 * Added support for 16khz signed linear media streams.
2196 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2197 RTP has been outfitted with the same abilities.
2198 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2199 available in device configurations as well as in the dial plan.
2200 * Addition of the 'subscribe_network_change' option for turning on and off
2201 res_stun_monitor module support in chan_sip.
2202 * Addition of the 'auth_options_requests' option for turning on and off
2203 authentication for OPTIONS requests in chan_sip.
2207 * Add #tryinclude statement for config files. This provides the same
2208 functionality as the #include statement however an asterisk module will
2209 still load if the filename does not exist. Using the #include statement
2210 Asterisk will not allow the module to load.
2214 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2215 on realtime updates.
2216 * Added the ability for chan_iax2 to inform the dialplan whether or not
2217 encryption is being used. This interoperates with the SIP SRTP implementation
2218 so that a secure SIP call can be bridged to a secure IAX call when the
2219 dialplan requires bridged channels to be "secure".
2220 * Addition of the 'subscribe_network_change' option for turning on and off
2221 res_stun_monitor module support in chan_iax.
2226 * Added ability to preset channel variables on indicated lines with the setvar
2227 configuration option. Also, clearvars=all resets the list of variables back
2229 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2230 See configs/res_pktccops.conf for more information.
2232 XMPP Google Talk/Jingle changes
2233 -------------------------------
2234 * Added the externip option to gtalk.conf.
2235 * Added the stunaddr option to gtalk.conf which allows for the automatic
2236 retrieval of the external ip from a stun server.
2240 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2241 match to a partial channel name.
2242 * Added .m3u support for Mp3Player application.
2243 * Added progress option to the app_dial D() option. When progress DTMF is
2244 present, those values are sent immediately upon receiving a PROGRESS message
2245 regardless if the call has been answered or not.
2246 * Added functionality to the app_dial F() option to continue with execution
2247 at the current location when no parameters are provided.
2248 * Added the 'a' option to app_dial to answer the calling channel before any
2249 announcements or macros are executed.
2250 * Modified app_dial to set answertime when the called channel answers even if
2251 the called channel hangs up during playback of an announcement.
2252 * Modified app_dial 'r' option to support an additional parameter to play an
2253 indication tone from indications.conf
2254 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2255 to cycle through the next available channel. By default this is still '*'.
2256 * Added x() option to app_chanspy. This option allows DTMF to be set to
2257 exit the application.
2258 * The Voicemail application has been improved to automatically ignore messages
2259 that only contain silence.
2260 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2261 associated mailbox(es) to be greetings-only.
2262 * The ChanSpy application now has the 'S' option, which makes the application
2263 automatically exit once it hits a point where no more channels are available
2265 * The ChanSpy application also now has the 'E' option, which spies on a single
2266 channel and exits when that channel hangs up.
2267 * The MeetMe application now turns on the DENOISE() function by default, for
2268 each participant. In our tests, this has significantly decreased background
2269 noise (especially noisy data centers).
2270 * Voicemail now permits storage of secrets in a separate file, located in the
2271 spool directory of each individual user. The control for this is located in
2272 the "passwordlocation" option in voicemail.conf. Please see the sample
2273 configuration for more information.
2274 * The ChanIsAvail application now exposes the returned cause code using a separate
2275 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2276 * Added 'd' option to app_followme. This option disables the "Please hold"
2278 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2279 received will terminate recording.
2280 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2281 Previously the folder could only be set per context, but has now been extended
2282 using the imapfolder option.
2283 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2284 * Voicemail now allows the pager date format to be specified separately from the
2286 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2287 to allow joining, leaving, and sending text to group chats.
2288 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2289 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2290 to all paged phones (and optionally excluding the caller's one using the new
2291 option 'n') before the call is bridged.
2292 * The 'f' option to Dial has been augmented to take an optional argument. If no
2293 argument is provided, the 'f' option works as it always has. If an argument is
2294 provided, then the connected party information of all outgoing channels created
2295 during the Dial will be set to the argument passed to the 'f' option.
2296 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2298 * The OSP lookup application adds in/outbound network ID, optional security,
2299 number portability, QoS reporting, destination IP port, custom info and service
2301 * Added new application VMSayName that will play the recorded name of the voicemail
2302 user if it exists, otherwise will play the mailbox number.
2303 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2304 retrieve state for a particular bridge, where <name> is the conference name
2305 * app_directory now allows exiting at any time using the operator or pound key.
2306 * Voicemail now supports setting a locale per-mailbox.
2307 * Two new applications are provided for declining counting phrases in multiple
2308 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2310 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2312 * Voicemail now includes rdnis within msgXXXX.txt file.
2313 * ExternalIVR now supports IPv6 addresses.
2314 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2315 at https://wiki.asterisk.org/wiki/x/oQBB
2316 * ParkedCall and Park can now specify the parking lot to use.
2320 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2321 over SRV records associated with a specific service. From the CLI, type
2322 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2323 details on how these may be used.
2324 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2325 pitch of a channel's tx and rx audio streams.
2326 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2327 setting various connected line and redirecting party information.
2328 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2329 support ISDN subaddressing.
2330 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2331 * For DAHDI channels, the CHANNEL() dialplan function now allows
2332 the dialplan to request changes in the configuration of the active
2333 echo canceller on the channel (if any), for the current call only.
2336 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2338 The possible values are:
2340 on - normal mode (the echo canceller is actually reinitialized)
2342 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2344 voice - voice mode (returns from FAX mode, reverting the changes that
2345 were made when FAX mode was requested)
2346 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2347 and setting variables on the channel which created the current channel.
2348 Administrators should take care to avoid naming conflicts, when multiple
2349 channels are dialled at once, especially when used with the Local channel
2350 construct (which all could set variables on the master channel). Usage
2351 of the HASH() dialplan function, with the key set to the name of the slave
2352 channel, is one approach that will avoid conflicts.
2353 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2355 * func_odbc now allows multiple row results to be retrieved without using
2356 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2357 from the same query by using the name of the function which retrieved the
2358 first row as an argument to ODBC_FETCH().
2359 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2360 dialplan. This function returns the content of the received message.
2361 * Added REPLACE, which searches a given variable name for a set of characters,
2362 then either replaces them with a single character or deletes them.
2363 * Added PASSTHRU, which literally passes the same argument back as its return
2364 value. The intent is to be able to use a literal string argument to
2365 functions that currently require a variable name as an argument.
2366 * HASH-associated variables now can be inherited across channel creation, by
2367 prefixing the name of the hash at assignment with the appropriate number of
2368 underscores, just like variables.
2369 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2370 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2371 whether or not channels that are bridged to the current channel will be
2372 required to have secure signaling and/or media.
2373 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2374 the current channel has secure signaling and/or media.
2375 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2376 "no_media_path" option.
2377 Returns "0" if there is a B channel associated with the call.
2378 Returns "1" if no B channel is associated with the call. The call is either
2379 on hold or is a call waiting call.
2380 * Added option to dialplan function CDR(), the 'f' option
2381 allows for high resolution times for billsec and duration fields.
2382 * FILE() now supports line-mode and writing.
2383 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2384 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2388 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2389 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2390 and is set when a dynamic feature is triggered.
2391 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2392 to dynamically create a new parking lot matching the value this varible is
2394 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2395 features.conf that should be the base for dynamic parkinglots.
2396 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2397 parkinglot should have.
2398 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2399 parkinglot should have.
2400 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2405 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2406 timeout has expired.
2407 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2408 to the caller when an Agent's phone is ringing. This can be used to indicate
2409 to the caller that their call is about to be picked up, which is nice when
2410 one has been on hold for an extened period of time.
2411 * A new config option, penaltymemberslimit, has been added to queues.conf.
2412 When set this option will disregard penalty settings when a queue has too
2414 * A new option, 'I' has been added to both app_queue and app_dial.
2415 By setting this option, Asterisk will not update the caller with
2416 connected line changes or redirecting party changes when they occur.
2417 * A 'relative-periodic-announce' option has been added to queues.conf. When
2418 enabled, this option will cause periodic announce times to be calculated
2419 from the end of announcements rather than from the beginning.
2420 * The autopause option in queues.conf can be passed a new value, "all." The
2421 result is that if a member becomes auto-paused, he will be paused in all
2422 queues for which he is a member, not just the queue that failed to reach
2424 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2425 * The queue logger now allows events to optionally propagate to a file,
2426 even when realtime logging is turned on. Additionally, realtime logging
2427 supports sending the event arguments to 5 individual fields, although it
2428 will fallback to the previous data definition, if the new table layout is
2431 mISDN channel driver (chan_misdn) changes
2432 ----------------------------------------
2433 * Added display_connected parameter to misdn.conf to put a display string
2434 in the CONNECT message containing the connected name and/or number if
2435 the presentation setting permits it.
2436 * Added display_setup parameter to misdn.conf to put a display string
2437 in the SETUP message containing the caller name and/or number if the
2438 presentation setting permits it.
2439 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2440 indicate the dialplan settings are to be obtained from the asterisk
2442 * Made misdn.conf parameter callerid accept the "name" <number> format
2443 used by the rest of the system.
2444 * Made use the nationalprefix and internationalprefix misdn.conf
2445 parameters to prefix any received number from the ISDN link if that
2446 number has the corresponding Type-Of-Number. NOTE: This includes
2447 comparing the incoming call's dialed number against the MSN list.
2448 * Added the following new parameters: unknownprefix, netspecificprefix,
2449 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2450 received number from the ISDN link if that number has the corresponding
2452 * Added new dialplan application misdn_command which permits controlling
2453 the CCBS/CCNR functionality.
2454 * Added new dialplan function mISDN_CC which permits retrieval of various
2455 values from an active call completion record.
2456 * For PTP, you should manually send the COLR of the redirected-to party
2457 for an incomming redirected call if the incoming call could experience
2458 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2459 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2460 if the REDIRECTING(from-num) is not empty.
2461 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2462 option on all of the REDIRECTING statements before dialing the
2463 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2464 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2465 redirecting-to presentation (COLR) when it becomes available.
2466 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2469 thirdparty mISDN enhancements
2470 -----------------------------
2471 mISDN has been modified by Digium, Inc. to greatly expand facility message
2473 * Enhanced COLP support for call diversion and transfer.
2474 * CCBS/CCNR support.
2476 The latest modified mISDN v1.1.x based version is available at:
2477 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2478 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2480 Tagged versions of the modified mISDN code are available under:
2481 http://svn.digium.com/svn/thirdparty/mISDN/tags
2482 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2484 libpri channel driver (chan_dahdi) DAHDI changes
2485 -------------------------------------------
2486 * The channel variable PRIREDIRECTREASON is now just a status variable
2487 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2488 to read and alter the reason.
2489 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2490 redirected-to party for an incomming redirected call if the incoming call
2491 could experience further redirects. Just set the
2492 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2493 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2495 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2496 use the inhibit(i) option on all of the REDIRECTING statements before
2497 dialing the redirected-to party. You still have to set the
2498 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2499 will update the redirecting-to presentation (COLR) when it becomes available.
2500 * Added the ability to ignore calls that are not in a Multiple Subscriber
2501 Number (MSN) list for PTMP CPE interfaces.
2502 * Added dynamic range compression support for dahdi channels. It is
2503 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2504 * Added support for ISDN calling and called subaddress with partial support
2505 for connected line subaddress.
2506 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2507 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2508 to transfer a held call on disconnect similar to an analog phone.
2509 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2510 Will reroute/deflect an outgoing call when receive the message.
2511 Can use the DAHDISendCallreroutingFacility to send the message for the
2513 * Added standard location to add options to chan_dahdi dialing:
2514 Dial(DAHDI/g1[/extension[/options]])
2517 R Reverse charging indication
2518 * Added Reverse Charging Indication (Collect calls) send/receive option.
2519 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2520 Dial(DAHDI/g1/extension/R)
2521 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2522 (requires latest LibPRI)
2523 * Added ability to send/receive keypad digits in the SETUP message.
2524 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2525 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2526 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2527 (requires latest LibPRI)
2528 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2529 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2530 back into the same interface. Tromboned calls happen because of call routing,
2531 call deflection, call forwarding, and call transfer.
2532 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2533 * Added the ability to support call waiting calls. (The SETUP has no B channel
2535 * Added Malicious Call ID (MCID) event to the AMI call event class.
2536 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2538 Asterisk Manager Interface
2539 --------------------------
2540 * The Hangup action now accepts a Cause header which may be used to
2541 set the channel's hangup cause.
2542 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2543 to specify a separate .pem file to hold a private key. By default sslcert
2544 is used to hold both the public and private key.
2545 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2546 for options containing the 'tls' prefix. For example, 'sslenable' is now
2547 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2548 across all .conf files. All affected sample.conf files have been modified to
2549 reflect this change. Previous options such as 'sslenable' still work,
2550 but options with the 'tls' prefix are preferred.
2551 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2552 in a channel. (res_mutestream.so)
2553 * The configuration file manager.conf now supports a channelvars option, which
2554 specifies a list of channel variables to include in each channel-oriented
2556 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2557 and ExtraPriority to allow redirecting the second channel to a different
2558 location than the first.
2559 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2561 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2562 in a MixMonitor recording.
2563 * The 'iax2 show peers' output is now similar to the expected output of
2565 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2567 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2568 AOC-E messages on a channel.
2569 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2570 conform more closely to similar events.
2571 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2573 * Added optional parkinglot variable for park command.
2574 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2575 if CallerIDNum and CallerIDName headers are also present.
2577 Channel Event Logging
2578 ---------------------
2579 * A new interface, CEL, is introduced here. CEL logs single events, much like
2580 the AMI, but it differs from the AMI in that it logs to db backends much
2581 like CDR does; is based on the event subsystem introduced by Russell, and
2582 can share in all its benefits; allows multiple backends to operate like CDR;
2583 is specialized to event data that would be of concern to billing sytems,
2584 like CDR. Backends for logging and accounting calls have been produced,
2585 but a new CDR backend is still in development.
2589 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2590 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2591 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2592 * Multiple files and formats can now be specified in cdr_custom.conf.
2593 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2594 See configs/cdr_syslog.conf.sample for more information.
2595 * A 'sequence' field has been added to CDRs which can be combined with
2596 linkedid or uniqueid to uniquely identify a CDR.
2597 * Handling of billsec and duration field has changed. If your table definition
2598 specifies those fields as float,double or similar they will now be logged with
2599 microsecond accuracy instead of a whole integer.
2601 Calendaring for Asterisk
2602 ------------------------
2603 * A new set of modules were added supporing calendar integration with Asterisk.
2604 Dialplan functions for reading from and writing to calendars are included,
2605 as well as the ability to execute dialplan logic upon calendar event notifications.
2606 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2607 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2608 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2609 2003 support does not support forms-based authentication).
2611 Call Completion Supplementary Services for Asterisk
2612 ---------------------------------------------------
2613 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2614 DAHDI/ISDN supports call completion for the following switch types:
2615 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2616 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2618 Multicast RTP Support
2619 ---------------------
2620 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2621 The channel driver can be used with the Page application to perform multicast RTP
2622 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2623 Type can be either basic or linksys.
2624 Destination is the IP address and port for the RTP packets.
2625 Control address is specific to the linksys type and is used for sending the control
2626 packets unique to them.
2628 Security Events Framework
2629 -------------------------
2630 * Asterisk has a new C API for reporting security events. The module res_security_log
2631 sends these events to the "security" logger level. Currently, AMI is the only
2632 Asterisk component that reports security events. However, SIP support will be
2633 coming soon. For more information on the security events framework, see the
2634 "Asterisk Security Framework" section of the Asterisk wiki at
2635 https://wiki.asterisk.org/wiki/x/wgBQ
2636 * SIP support was added in Asterisk 10
2637 * This API now supports IPv6 addresses
2641 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2642 * A spandsp based fax backend (res_fax_spandsp) has been added.
2643 * The app_fax module has been deprecated in favor of the res_fax module and
2644 the new res_fax_spandsp backend.
2645 * The SendFAX and ReceiveFAX applications now send their log messages to a
2646 'fax' logger level, instead of to the generic logger levels. To see these
2647 messages, the system's logger.conf file will need to direct the 'fax' logger
2648 level to one or more destinations; the logger.conf.sample file includes an
2649 example of how to do this. Note that if the 'fax' logger level is *not*
2650 directed to at least one destination, log messages generated by these
2651 applications will be lost, and that if the 'fax' logger level is directed to
2652 the console, the 'core set verbose' and 'core set debug' CLI commands will
2653 have no effect on whether the messages appear on the console or not.
2657 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2658 Now, in order to enable transmitting silence during record the transmit_silence
2659 option should be used. transmit_silence_during_record remains a valid option, but
2660 defaults to the behavior of the transmit_silence option.
2661 * Addition of the Unit Test Framework API for managing registration and execution
2662 of unit tests with the purpose of verifying the operation of C functions.
2663 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2664 XMPP text messages to the remote JID.
2665 * Modules.conf has a new option - "require" - that marks a module as critical for
2666 the execution of Asterisk.
2667 If one of the required modules fail to load, Asterisk will exit with a return
2669 * An 'X' option has been added to the asterisk application which enables #exec support.
2670 This allows #exec to be used in asterisk.conf.
2671 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2672 * A new lockconfdir option has been added to asterisk.conf to protect the
2673 configuration directory (/etc/asterisk by default) during reloads.
2674 * The parkeddynamic option has been added to features.conf to enable the creation
2675 of dynamic parkinglots.
2676 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2677 the reportalarms config option.
2678 * chan_dahdi supports dialing configuring and dialing by device file name.
2679 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2680 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2681 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2682 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2683 Handy for the above name-based syntax as it does not depend on
2684 initialization order.
2685 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2686 significant increase in performance (about 3X) for installations using this switchtype.
2687 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2688 AIS. For more information, please see the Distributed Device State section of the
2689 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2690 * The addition of G.719 pass-through support.
2691 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2692 during device configuration.
2693 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2694 have less than 3 lines on the LCD.
2695 * Realtime now supports database failover. See the sample extconfig.conf for details.
2696 * The addition of improved translation path building for wideband codecs. Sample
2697 rate changes during translation are now avoided unless absolutely necessary.
2698 * The addition of the res_stun_monitor module for monitoring and reacting to network
2699 changes while behind a NAT.
2700 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2701 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2702 These allow support for any Administration. Default is AT&T values.
2706 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2707 optionally accept a filename, to apply the setting only to the code generated from
2708 that source file when Asterisk was built. However, there are some modules in Asterisk
2709 that are composed of multiple source files, so this did not result in the behavior
2710 that users expected. In this version, 'core set debug' and 'core set verbose'
2711 can optionally accept *module* names instead (with or without the .so extension),
2712 which applies the setting to the entire module specified, regardless of which source
2713 files it was built from.
2714 * New 'manager show settings' command showing the current settings loaded from
2716 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2717 the channel hangup request to all channels.
2718 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2720 ------------------------------------------------------------------------------
2721 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2722 ------------------------------------------------------------------------------
2726 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2727 Snom phones use this for call pickup of extensions that the phone is
2729 * Added support for setting the domain in the URI for caller of an
2730 outbound call by using the SIPFROMDOMAIN channel variable.
2731 * Added a new configuration option "remotesecret" for authentication to
2732 remote services. For backwards compatibility, "secret" still has the
2733 same function as before, but now you can configure both a remote secret and a
2734 local secret for mutual authentication.
2735 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2736 the sound will be played to the target of an attended transfer
2737 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2738 finer control over how many peers Asterisk will qualify and the gap between them
2739 when all peers need to be qualified at the same time.
2740 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2741 (either globally or for a specific peer), chan_sip will treat any SDP data
2742 it receives as new data and update the media stream accordingly. By
2743 default, Asterisk will only modify the media stream if the SDP session
2744 version received is different from the current SDP session version. This
2745 option is required to interoperate with devices that have non-standard SDP
2746 session version implementations (observed with Microsoft OCS). This option
2747 is disabled by default.
2748 * The parsing of register => lines in sip.conf has been modified to allow a port
2749 to be present in the "user" portion. Please see the sip.conf.sample file for more
2751 * Added support for subscribing to MWI on a remote server and making the status available
2752 as a mailbox. Please see the sip.conf.sample file for more information.
2753 * Added a function to remove SIP headers added in the dialplan before the
2754 first INVITE is generated - SIPRemoveHeader()
2755 * Channel variables set with setvar= in a device configuration is now
2756 set both for inbound and outbound calls.
2757 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2761 * Added immediate option to iax.conf
2762 * Added forceencryption option to iax.conf
2763 * Added Encryption and Trunk status to manager command "iaxpeers"
2767 * The configuration file now holds separate sections for devices and lines.
2768 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2773 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2774 support for LibOpenR2. http://www.libopenr2.org/
2775 * The UK option waitfordialtone has been added for use with BT analog
2777 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2778 is used in conjunction with the 'faxdetect' configuration option. When
2779 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2780 switch to the configured faxbuffers policy. For example, to use 6 buffers
2781 and a 'full' buffer policy for a fax transmission, add:
2783 The faxbuffers configuration will be in affect until the call is torn down.
2784 * Added service message support for 4ESS/5ESS switches.
2788 * For DAHDI channels, the CHANNEL() dialplan function now
2789 supports changing the channel's buffer policy (for the current
2790 call only), using this syntax:
2792 exten => s,n,Set(CHANNEL(buffers)=6,full)
2794 This would change the channel to the 'full' buffer policy and
2795 6 (six) buffers. Possible options for this setting are the same
2796 as those in chan_dahdi.conf.
2797 * Added a new dialplan function, CURLOPT, which permits setting various
2798 options that may be useful with the CURL dialplan function, such as
2799 cookies, proxies, connection timeouts, passwords, etc.
2800 * Permit the syntax and synopsis fields of the corresponding dialplan
2801 functions to be individually set from func_odbc.conf.
2802 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2803 * func_odbc now may specify an insert query to execute, when the write query
2804 affects 0 rows (usually indicating that no such row exists).
2805 * Added a new dialplan function, LISTFILTER, which permits removing elements
2806 from a set list, by name. Uses the same general syntax as the existing CUT
2807 and FIELDQTY dialplan functions, which also manage lists.
2808 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2809 obtaining realtime data from the dialplan.
2810 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2811 a subroutine when using the GoSub() and Return() applications.
2812 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2813 of "core show function AUDIOHOOK_INHERIT" from the CLI
2814 * Added AES_ENCRYPT. For information on its use, please see the output
2815 of "core show function AES_ENCRYPT" from the CLI
2816 * Added AES_DECRYPT. For information on its use, please see the output
2817 of "core show function AES_DECRYPT" from the CLI
2818 * func_odbc now supports database transactions across multiple queries.
2822 * Scheduled meetme conferences may now have their end times extended by
2824 * app_authenticate now gives the ability to select a prompt other than
2826 * app_directory now pays attention to the searchcontexts setting in
2827 voicemail.conf and will look through all contexts, if no context is
2828 specified in the initial argument.
2829 * A new application, Originate, has been introduced, that allows asynchronous
2830 call origination from the dialplan.
2831 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2832 in addition to the setting in the "general" context.
2833 * Added ConfBridge dialplan application which does conference bridges without
2834 DAHDI. For information on its use, please see the output of
2835 "core show application ConfBridge" from the CLI.
2839 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2840 operation to the AMI Redirect action.
2841 * extensions.conf now allows you to use keyword "same" to define an extension
2842 without actually specifying an extension. It uses exactly the same pattern
2843 as previously used on the last "exten" line. For example:
2844 exten => 123,1,NoOp(something)
2845 same => n,SomethingElse()
2846 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2847 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2848 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2849 by the new clialiases module. See cli_aliases.conf.sample file.
2850 * Times within timespecs are now accurate down to the minute. This is a change
2851 from historical Asterisk, which only provided timespecs rounded to the nearest
2852 even (read: evenly divisible by 2) minute mark.
2853 * The realtime switch now supports an option flag, 'p', which disables searches for
2855 * In addition to a time range and date range, timespecs now accept a 5th optional
2856 argument, timezone. This allows you to perform time checks on alternate
2857 timezones, especially if those daylight savings time ranges vary from your
2858 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2860 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2861 give you the correct output for an asterisk box behind nat. It will give you the
2862 externhost and localnet settings.
2863 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2864 can connect calls in passthrough mode, as well as record and play back files.
2865 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2866 using pickupsound and pickupfailsound in features.conf.
2867 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2868 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2869 instead of the /var/run/asterisk.pid where it used to be. This will make
2870 installs as non-root easier to manage.
2875 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2876 be written; they will no longer be explicitly written.
2878 Asterisk Manager Interface
2879 --------------------------
2880 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2881 a non-empty value) in your request. If you do this, any pending AMI events will
2882 *not* be included in the response to your request as they would normally, but
2883 will be left in the event queue for the next request you make to retrieve. For
2884 some applications, this will allow you to guarantee that you will only see
2885 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2886 To know whether the Asterisk server supports this header or not, your client can
2887 inspect the first response back from the server to see if it includes this header:
2889 Pragma: SuppressEvents
2891 If this is included, the server supports event suppression.
2893 * Added 4 new Actions to list skinny device(s) and line(s)
2899 LDAP Schema File Additions
2900 --------------------------
2901 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2902 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2904 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2905 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2906 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2907 * Removed redundant IPaddr (there's already IPAddress)
2908 - Gives more configuration Flags for SIP-Users available (tested)
2909 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2910 without extensibleObject (which really should be the last resort); gives
2911 also additional possibilities for LDAP-filter
2913 ------------------------------------------------------------------------------
2914 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2915 ------------------------------------------------------------------------------
2917 Device State Handling
2918 ---------------------
2919 * The event infrastructure in Asterisk got another big update to help support
2920 distributed events. It currently supports distributed device state and
2921 distributed Voicemail MWI (Message Waiting Indication). A new module has
2922 been merged, res_ais, which facilitates communicating events between servers.
2923 It uses the SAForum AIS (Service Availability Forum Application Interface
2924 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2925 a cluster of Asterisk servers, and to share events between them. For more
2926 information on setting this up, refer to the Distributed Device State section
2927 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2931 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2932 variables from an Asterisk configuration file.
2933 * The JACK_HOOK function now has a c() option to supply a custom client name.
2934 * Added two new dialplan functions from libspeex for audio gain control and
2935 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2936 rx directions of a channel from the dialplan.
2937 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2938 based on other parameters. The default is still to search based on the
2939 forwarding station ID. However, there are new options that allow you to search
2940 based on the message desk terminal ID, or the message desk number.
2941 * TIMEOUT() has been modified to be accurate down to the millisecond.
2942 * ENUM*() functions now include the following new options:
2943 - 'u' returns the full URI and does not strip off the URI-scheme.
2944 - 's' triggers ISN specific rewriting
2945 - 'i' looks for branches into an Infrastructure ENUM tree
2946 - 'd' for a direct DNS lookup without any flipping of digits.
2947 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2948 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2949 deviation of jitter, rtt, and loss for a call using chan_sip.
2951 DAHDI channel driver (chan_dahdi) Changes
2952 ----------------------------------------
2953 * Channels can now be configured using named sections in chan_dahdi.conf, just
2954 like other channel drivers, including the use of templates.
2955 * The default for pridialplan has changed from 'national' to 'unknown'.
2959 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2960 to something that matches the pattern a hint will be created using the contents
2961 and variables evaluated.
2962 * Dialplan matching has been extended to allow an extension to return to the
2963 PBX core to wait for more digits. This is done by using the new dialplan
2964 application called "Incomplete". This will permit a whole new level of
2965 extension control, by giving the administrator more control over early
2966 matches employing one of the short-circuit pattern match operators. Note
2967 that custom applications can trigger this same behavior by returning the
2968 special value AST_PBX_INCOMPLETE.
2972 * Directory now permits both first and last names to be matched at the same
2973 time. In addition, the number of digits to enter of the name can be set in
2974 the arguments to Directory; previously, you could enter only 3, regardless
2975 of how many names are in your company. For large companies, this should be
2977 * Voicemail now permits a mailbox setting to wrap around from first to last
2978 messages, if the "messagewrap" option is set to a true value.
2979 * Voicemail now permits an external script to be run, for password validation.
2980 The script should output "VALID" or "INVALID" on stdout, depending upon the
2981 wish to validate or invalidate the password given. Arguments are:
2982 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2984 * Dial has a new option: F(context^extension^pri), which permits a callee to
2985 continue in the dialplan, at the specified label, if the caller hangs up.
2986 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2987 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2988 * The Jack application now has a c() option to supply a custom client name.
2989 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2990 like the pre-existing whisper mode, except that the spy can also talk to the
2991 participant on the bridged channel as well.
2992 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
2993 to be spoken instead of the channel name or number. For more information on the
2994 use of this option, issue the command "core show application ChanSpy" from the
2996 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
2997 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
2998 words, if using the 'd' option, it is not possible to enter a number to append to
2999 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3000 change to whisper mode, and pressing 6 will change to barge mode.
3001 * ExternalIVR now takes several options that affect the way it performs, as
3002 well as having several new commands. Please see the External IVR page on the Asterisk
3003 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3004 * Added ability to communicate over a TCP socket instead of forking a child process for the
3005 ExternalIVR application.
3006 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3007 of just the first one if you give the function more then one channel to check.
3008 * PrivacyManager now takes an option where you can specify a context where the
3009 given number will be matched. This way you have more control over who is allowed
3010 and it stops the people who blindly enter 10 digits.
3011 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3012 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3013 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3014 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3015 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3016 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3017 * The Dial() application no longer copies the language used by the caller to the callee's
3018 channel. If you desire for the caller's channel's language to be used for file playback
3019 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3020 * SendImage() no longer hangs up the channel on error; instead, it sets the
3021 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3022 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3024 * Park has a new option, 's', which silences the announcement of the parking space number.
3025 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3026 invalid input and will be assumed to mean that no timeout is desired.
3030 * Added DNS manager support to registrations for peers referencing peer entries.
3031 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3032 as well as periodically updating the IP address. These properties allow for
3033 better performance as well as recovery in the event of an IP change.
3034 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3035 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3036 These changes also provide performance improvements for call setup and tear down.
3037 * Added ability to specify registration expiry time on a per registration basis in
3039 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3041 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3042 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3043 * 'sip show peers' and 'sip show users' display their entries sorted in
3044 alphabetical order, as opposed to the order they were in, in the config
3046 * Videosupport now supports an additional option, "always", which always sets
3047 up video RTP ports, even on clients that don't support it. This helps with
3048 callfiles and certain transfers to ensure that if two video phones are
3049 connected, they will always share video feeds.
3053 * Existing DNS manager lookups extended to check for SRV records.
3054 * IAX2 encryption support has been improved to support periodic key rotation
3055 within a call for enhanced security. The option "keyrotate" has been
3056 provided to disable this functionality to preserve backwards compatibility
3057 with older versions of IAX2 that do not support key rotation.
3061 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3062 data tree based on the given <path>.
3063 * New CLI command "data show providers" that will display all the registered
3065 * New CLI command, "config reload <file.conf>" which reloads any module that
3066 references that particular configuration file. Also added "config list"
3067 which shows which configuration files are in use.
3068 * New CLI commands, "pri show version" and "ss7 show version" that will
3069 display which version of libpri and libss7 are being used, respectively.
3070 A new API call was added so trunk will now have to be compiled against
3071 a versions of libpri and libss7 that have them or it will not know that
3072 these libraries exist.
3073 * The commands "core show globals", "core set global" and "core set chanvar" has
3074 been deprecated in favor of the more semanticly correct "dialplan show globals",
3075 "dialplan set chanvar" and "dialplan set global".
3076 * New CLI command "dialplan show chanvar" to list all variables associated
3077 with a given channel.
3081 * Addresses managed by DNS manager now can check to see if there is a DNS
3082 SRV record for a given domain and will use that hostname/port if present.
3084 AMI - The manager (TCP/TLS/HTTP)
3085 --------------------------------
3086 * The Status command now takes an optional list of variables to display
3087 along with channel status.
3088 * The QueueEntry event now also includes the channel's uniqueid
3092 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3093 as some people were running into this limit. This limit has been increased
3098 * The TRANSFER queue log entry now includes the the caller's original
3099 position in the transferred-from queue.
3100 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3101 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3102 as well as an explanation about timeout options in general
3103 * Added a new option - C - for forcing the "answered elsewhere" flag on
3104 cancellation of calls in to members of the queue. This is to avoid the
3105 call to a member of a queue having the call listed as a "missed call".
3109 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3110 adaptive capabilities. What this means in practical terms is that if your
3111 realtime table lacks critical fields, Asterisk will now emit warnings to
3112 that effect. Also, some of the realtime drivers have the ability (if
3113 configured) to automatically add those columns to the table with the
3114 correct type and length.
3118 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3119 the 'setvar' option to cause a given audio file to be played upon completion
3120 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3121 Skinny channels only.
3122 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3123 Hoard page on the Asterisk wiki for more information:
3124 https://wiki.asterisk.org/wiki/x/pQBB
3125 * Config file variables may now be appended to, by using the '+=' append
3126 operator. This is most helpful when working with long SQL queries in
3127 func_odbc.conf, as the queries no longer need to be specified on a single
3129 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3130 which will add a second to the billsec when the ending
3131 time is set, if the number in the microseconds field of the end time is
3132 greater than the number of microseconds in the answer time. This allows
3133 users to count the 'initiated' seconds in their billing records.
3135 ------------------------------------------------------------------------------
3136 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3137 ------------------------------------------------------------------------------
3139 AMI - The manager (TCP/TLS/HTTP)
3140 --------------------------------
3141 * Manager has undergone a lot of changes, all of them documented
3142 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3143 * Manager version has changed to 1.1
3144 * Added a new action 'CoreShowChannels' to list currently defined channels
3145 and some information about them.
3146 * Added a new action 'SIPshowregistry' to list SIP registrations.
3147 * Added TLS support for the manager interface and HTTP server
3148 * Added the URI redirect option for the built-in HTTP server
3149 * The output of CallerID in Manager events is now more consistent.
3150 CallerIDNum is used for number and CallerIDName for name.
3151 * Enable https support for builtin web server.
3152 See configs/http.conf.sample for details.
3153 * Added a new action, GetConfigJSON, which can return the contents of an
3154 Asterisk configuration file in JSON format. This is intended to help
3155 improve the performance of AJAX applications using the manager interface
3157 * SIP and IAX manager events now use "ChannelType" in all cases where we
3158 indicate channel driver. Previously, we used a mixture of "Channel"
3159 and "ChannelDriver" headers.
3160 * Added a "Bridge" action which allows you to bridge any two channels that
3161 are currently active on the system.
3162 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3163 the voicemail users setup.
3164 * Added 'DBDel' and 'DBDelTree' manager commands.
3165 * cdr_manager now reports events via the "cdr" level, separating it from
3166 the very verbose "call" level.
3167 * Manager users are now stored in memory. If you change the manager account
3168 list (delete or add accounts) you need to reload manager.
3169 * Added Masquerade manager event for when a masquerade happens between
3171 * Added "manager reload" command for the CLI
3172 * Lots of commands that only provided information are now allowed under the
3173 Reporting privilege, instead of only under Call or System.
3174 * The IAX* commands now require either System or Reporting privilege, to
3175 mirror the privileges of the SIP* commands.
3176 * Added ability to retrieve list of categories in a config file.
3177 * Added ability to retrieve the content of a particular category.
3178 * Added ability to empty a context.
3179 * Created new action to create a new file.
3180 * Updated delete action to allow deletion by line number with respect to category.
3181 * Added new action insert to add new variable to category at specified line.
3182 * Updated action newcat to allow new category to be inserted in file above another
3184 * Added new event "JitterBufStats" in the IAX2 channel
3185 * Originate now requires the Originate privilege and, if you want to call out
3186 to a subshell, it requires the System privilege, as well. This was done to
3187 enhance manager security.
3188 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3189 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3190 or manager show command Atxfer from the CLI
3191 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3192 details or manager show command IAXregistry from the CLI
3196 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3197 state in the dialplan, as well as creating custom device states that are
3198 controllable from the dialplan.
3199 * Extend CALLERID() function with "pres" and "ton" parameters to
3200 fetch string representation of calling number presentation indicator
3201 and numeric representation of type of calling number value.
3202 * MailboxExists converted to dialplan function
3203 * A new option to Dial() for telling IP phones not to count the call
3204 as "missed" when dial times out and cancels.
3205 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3206 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3207 held for any given channel. Also, locks are automatically freed when a
3209 * Added HINT() dialplan function that allows retrieving hint information.
3210 Hints are mappings between extensions and devices for the sake of
3211 determining the state of an extension. This function can retrieve the list
3212 of devices or the name associated with a hint.
3213 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3215 * Added SYSINFO() dialplan function which allows retrieval of system information
3216 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3217 the existence of a dialplan target.
3218 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3219 upper and lower case, respectively.
3220 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3221 ID for the call (not the Asterisk call ID or unique ID), provided that the
3222 channel driver supports this. For SIP, you get the SIP call-ID for the
3223 bridged channel which you can store in the CDR with a custom field.
3227 * Added CLI permissions, config file: cli_permissions.conf
3228 default is to allow all commands for every local user/group.
3229 Also this new feature added three new CLI commands:
3230 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3231 - cli reload permissions
3232 - cli show permissions
3233 * New CLI command "core show hint" (usage: core show hint <exten>)
3234 * New CLI command "core show settings"
3235 * Added 'core show channels count' CLI command.
3236 * Added the ability to set the core debug and verbose values on a per-file basis.
3237 * Added 'queue pause member' and 'queue unpause member' CLI commands
3238 * Ability to set process limits ("ulimit") without restarting Asterisk
3239 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3240 output to make debugging on busy systems much easier.
3241 * New CLI commands "dialplan set extenpatternmatching true/false"
3242 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3243 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3244 listed in the startup_commands section of cli.conf will get executed.
3245 * Added a CLI command, "devstate change", which allows you to set custom device
3246 states from the func_devstate module that provides the DEVICE_STATE() function
3247 and handling of the "Custom:" devices.
3248 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3249 sorted into the different possible callbacks, with the number of entries
3250 currently scheduled for each. Gives you a feel for how busy the sip channel
3252 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3253 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3254 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3258 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3259 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3260 for a received call. If it is detected, the channel will jump to the
3261 'fax' extension in the dialplan.
3262 * The default SIP useragent= identifier now includes the Asterisk version
3263 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3264 If set, and the incoming request carries authentication info,
3265 the username to match in the users list is taken from the Digest header
3266 rather than from the From: field. This feature is considered experimental.
3267 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3268 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3269 * The "localmask" setting was removed in version 1.2 and the reminder about it
3270 being removed is now also removed.
3271 * A new option "busylevel" for setting a level of calls where asterisk reports
3272 a device as busy, to separate it from call-limit. This value is also added
3273 to the SIP_PEER dialplan function.
3274 * A new realtime family called "sipregs" is now supported to store SIP registration
3275 data. If this family is defined, "sippeers" will be used for configuration and
3276 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3277 registration data, as before.
3278 * The SIPPEER function have new options for port address, call and pickup groups
3279 * Added support for T.140 realtime text in SIP/RTP
3280 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3281 required due to the restructuring of how MWI is handled. See the descriptions
3282 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3283 for more information.
3284 * Added rtpdest option to CHANNEL() dialplan function.
3285 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3286 * SIP now adds a header to the CANCEL if the call was answered by another phone
3287 in the same dial command, or if the new c option in dial() is used.
3288 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3289 states it is not needed. For phones, however, that do require it the "registertrying" option
3290 has been added so it can be enabled.
3291 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3292 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3293 used to enable this functionality).
3294 * New settings for timer T1 and timer B on a global level or per device. This makes it
3295 possible to force timeout faster on non-responsive SIP servers. These settings are
3296 considered advanced, so don't use them unless you have a problem.
3297 * Added a dial string option to be able to set the To: header in an INVITE to any
3299 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3300 the qualify frequency.
3301 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3302 were not properly torn down due to network or endpoint failures during an established
3304 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3305 and configs/sip.conf.sample for more information on how it is used.
3306 * Added a new configuration option "authfailureevents" that enables manager events when
3307 a peer can't authenticate properly.
3308 * Added DNS manager support to registrations for peers not referencing a peer entry.
3312 * Added the trunkmaxsize configuration option to chan_iax2.
3313 * Added the srvlookup option to iax.conf
3314 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3317 XMPP Google Talk/Jingle changes
3318 -------------------------------
3319 * Added the bindaddr option to gtalk.conf.
3323 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
3324 * Proper codec support in chan_skinny.
3325 * Added settings for IP and Ethernet QoS requests
3329 * Added separate settings for media QoS in mgcp.conf
3331 Console Channel Driver changes
3332 ------------------------------
3333 * Added experimental support for video send & receive to chan_oss.
3334 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
3337 Phone channel changes (chan_phone)
3338 ----------------------------------
3339 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
3341 H.323 channel Changes
3342 ---------------------
3343 * H323 remote hold notification support added (by NOTIFY message
3344 and/or H.450 supplementary service)
3346 Local channel changes
3347 ---------------------
3348 * The device state functionality in the Local channel driver has been updated
3349 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
3350 to just UNKNOWN if the extension exists.
3351 * Added jitterbuffer support for chan_local. This allows you to use the
3352 generic jitterbuffer on incoming calls going to Asterisk applications.
3353 For example, this would allow you to use a jitterbuffer for an incoming
3354 SIP call to Voicemail by putting a Local channel in the middle. This
3355 feature is enabled by using the 'j' option in the Dial string to the Local
3356 channel in conjunction with the existing 'n' option for local channels.
3357 * A 'b' option has been added which causes chan_local to return the actual channel
3358 that is behind it when queried. This is useful for transfer scenarios as the
3359 actual channel will be transferred, not the Local channel.
3361 Agent channel changes
3362 ----------------------
3363 * The ackcall and endcall options are now supplemented with options acceptdtmf
3364 and enddtmf. These allow for the DTMF keypress to be configurable. The options
3365 default to their old hard-coded values ('#' and '*' respectively) so this should
3366 not break any existing agent installations.
3368 DAHDI channel driver (chan_dahdi) Changes
3369 ----------------------------------------
3370 * SS7 support (via libss7 library)
3371 * In India, some carriers transmit CID via dtmf. Some code has been added
3372 that will handle some situations. The cidstart=polarity_IN choice has been added for
3373 those carriers that transmit CID via dtmf after a polarity change.
3374 * CID matching information is now shown when doing 'dialplan show'.
3375 * Added dahdi show version CLI command.
3376 * Added setvar support to chan_dahdi.conf channel entries.
3377 * Added two new options: mwimonitor and mwimonitornotify. These options allow
3378 you to enable MWI monitoring on FXO lines. When the MWI state changes,
3379 the script specified in the mwimonitornotify option is executed. An internal
3380 event indicating the new state of the mailbox is also generated, so that
3381 the normal MWI facilities in Asterisk work as usual.
3382 * Added signalling type 'auto', which attempts to use the same signalling type
3383 for a channel as configured in DAHDI. This is primarily designed for analog
3384 ports, but will also work for digital ports that are configured for FXS or FXO
3385 signalling types. This mode is also the default now, so if your chan_dahdi.conf
3386 does not specify signalling for a channel (which is unlikely as the sample
3387 configuration file has always recommended specifying it for every channel) then
3388 the 'auto' mode will be used for that channel if possible.
3389 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
3390 state for a channel; also ensured that the DNDState Manager event is
3391 emitted no matter how the DND state is set or cleared.
3395 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
3396 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
3397 for details. This new channel driver allows you to use Nortel i2002,
3398 i2004, and i2050 phones with Asterisk.
3399 * Added a new channel driver, chan_console, which uses portaudio as a cross
3400 platform audio interface. It was written as a channel driver that would
3401 work with Mac CoreAudio, but portaudio supports a number of other audio
3402 interfaces, as well. Note that this channel driver requires v19 or higher
3403 of portaudio; older versions have a different API.
3407 * Added the ability to specify arguments to the Dial application when using
3408 the DUNDi switch in the dialplan.
3409 * Added the ability to set weights for responses dynamically. This can be
3410 done using a global variable or a dialplan function. Using the SHELL()
3411 function would allow you to have an external script set the weight for
3413 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
3414 functions will allow you to initiate a DUNDi query from the dialplan,
3415 find out how many results there are, and access each one.
3416 * Added the ability to specifiy a port for a dundi peer.
3420 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
3421 functions will allow you to initiate an ENUM lookup from the dialplan,
3422 and Asterisk will cache the results. ENUMRESULT can be used to access
3423 the results without doing multiple DNS queries.
3427 * Added the ability to customize which sound files are used for some of the
3428 prompts within the Voicemail application by changing them in voicemail.conf
3429 * Added the ability for the "voicemail show users" CLI command to show users
3430 configured by the dynamic realtime configuration method.
3431 * MWI (Message Waiting Indication) handling has been significantly
3432 restructured internally to Asterisk. It is now totally event based
3433 instead of polling based. The voicemail application will notify other
3434 modules that have subscribed to MWI events when something in the mailbox
3436 This also means that if any other entity outside of Asterisk is changing
3437 the contents of mailboxes, then the voicemail application still needs to
3438 poll for changes. Examples of situations that would require this option
3439 are web interfaces to voicemail or an email client in the case of using
3440 IMAP storage. So, two new options have been added to voicemail.conf
3441 to account for this: "pollmailboxes" and "pollfreq". See the sample
3442 configuration file for details.
projects / asterisk/asterisk.git / blob