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 * The ConfBridge dialplan application now sets a channel variable,
25 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
26 how a channel exited the conference.
28 * Added conference user option 'announce_join_leave_review'. This option
29 implies 'announce_join_leave' with the added effect that the user will
30 be asked if they want to confirm or re-record the recording of their
31 name when entering the conference
34 --------------------------
35 * At exit, the Directory application now sets a channel variable
36 DIRECTORY_RESULT to one of the following based on the reason for exiting:
37 OPERATOR user requested operator by pressing '0' for operator
38 ASSISTANT user requested assistant by pressing '*' for assistant
39 TIMEOUT user pressed nothing and Directory stopped waiting
40 HANGUP user's channel hung up
41 SELECTED user selected a user from the directory and is routed
42 USEREXIT user pressed '#' from the selection prompt to exit
43 FAILED directory failed in a way that wasn't accounted for. Dang.
46 --------------------------
47 * PickupChan now accepts channel uniqueids of channels to pickup.
50 --------------------------
51 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
52 to 'true' (case insensitive), then any Say application (SayNumber,
53 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
54 anticipate DTMF. If DTMF is received, these applications will behave like
55 the background application and jump to the received extension once a match
56 is established or after a short period of inactivity.
59 -------------------------
60 * A new function, MIXMONITOR, has been added to allow access to individual
61 instances of MixMonitor on a channel.
63 ------------------------------------------------------------------------------
64 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
65 ------------------------------------------------------------------------------
70 Asterisk 12 is a standard release of the Asterisk project. As such, the
71 focus of development for this release was on core architectural changes and
72 major new features. This includes:
73 * A more flexible bridging core based on the Bridging API
74 * A new internal message bus, Stasis
75 * Major standardization and consistency improvements to AMI
76 * Addition of the Asterisk RESTful Interface (ARI)
77 * A new SIP channel driver, chan_pjsip
78 In addition, as the vast majority of bridging in Asterisk was migrated to the
79 Bridging API used by ConfBridge, major changes were made to most of the
80 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
82 Specifications have been written for the affected interfaces. These
83 specifications are available on the Asterisk wiki:
84 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
85 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
86 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
88 It is *highly* recommended that anyone migrating to Asterisk 12 read the
89 information regarding its release both in this file and in the accompanying
90 UPGRADE.txt file. More detailed information on the major changes can be found
91 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
96 * Added build option DISABLE_INLINE. This option can be used to work around a
97 bug in gcc. For more information, see
98 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
100 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
101 the CHANNEL_TRACE build option were incompatible with the new bridging
104 * Asterisk now optionally uses libxslt to improve XML documentation generation
105 and maintainability. If libxslt is not available on the system, some XML
106 documentation will be incomplete.
108 * Asterisk now depends on libjansson. If a package of libjansson is not
109 available on your distro, please see http://www.digip.org/jansson/.
111 * Asterisk now depends on libuuid and, optionally, uriparser. It is
112 recommended that you install uriparser, even if it is optional.
114 * The new SIP stack and channel driver uses a particular version of PJSIP.
115 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
116 configuring and installing PJSIP for usage with Asterisk.
118 * Optional API was re-implemented to be more portable, and no longer requires
119 weak reference support from the compiler. The build option OPTIONAL_API may
120 be disabled to disable Optional API support.
127 * Along with AgentRequest, this application has been modified to be a
128 replacement for chan_agent. The act of a channel calling the AgentLogin
129 application places the channel into a pool of agents that can be
130 requested by the AgentRequest application. Note that this application, as
131 well as all other agent related functionality, is now provided by the
132 app_agent_pool module. See chan_agent and AgentRequest for more information.
134 * This application no longer performs agent authentication. If authentication
135 is desired, the dialplan needs to perform this function using the
136 Authenticate or VMAuthenticate application or through an AGI script before
139 * If this application is called and the agent is already logged in, the
140 dialplan will continue exection with the AGENT_STATUS channel variable set
141 to ALREADY_LOGGED_IN.
143 * The agents.conf schema has changed. Rather than specifying agents on a
144 single line in comma delineated fashion, each agent is defined in a separate
145 context. This allows agents to use the power of context templates in their
148 * A number of parameters from agents.conf have been removed. This includes
149 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
150 urlprefix, and savecallsin. These options were obsoleted by the move from
151 a channel driver model to the bridging/application model provided by
156 * A new application, this will request a logged in agent from the pool and
157 bridge the requested channel with the channel calling this application.
158 Logged in agents are those channels that called the AgentLogin application.
159 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
160 application will be set with an appropriate error value.
164 * This application has been removed. It was a holdover from when
165 AgentCallbackLogin was removed.
169 * Added support for additional Ademco DTMF signalling formats, including
170 Express 4+1, Express 4+2, High Speed and Super Fast.
172 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
173 call time, in milliseconds, to run the application.
175 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
176 maximum number of times to retry the call.
178 * Added a new configuration option answait. If set, the AlarmReceiver
179 application will wait the number of milliseconds specified by answait
180 after the channel has answered. Valid values range between 500
181 milliseconds and 10000 milliseconds.
183 * Added configuration option no_group_meta. If enabled, grouping of metadata
184 information in the AlarmReceiver log file will be skipped.
188 * A new application in Asterisk, this will place the calling channel
189 into a holding bridge, optionally entertaining them with some form of
190 media. Channels participating in a holding bridge do not interact with
191 other channels in the same holding bridge. Optionally, however, a channel
192 may join as an announcer. Any media passed from an announcer channel is
193 played to all channels in the holding bridge. Channels leave a holding
194 bridge either when an optional timer expires, or via the ChannelRedirect
195 application or AMI Redirect action.
199 * All participants in a bridge can now be kicked out of a conference room
200 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
201 command, i.e., 'confbridge kick <conference> all'
203 * CLI output for the 'confbridge list' command has been improved. When
204 displaying information about a particular bridge, flags will now be shown
205 for the participating users indicating properties of that user.
207 * The ConfbridgeList event now contains the following fields: WaitMarked,
208 EndMarked, and Waiting. This displays additional properties about the
209 user's profile, as well as whether or not the user is waiting for a
210 Marked user to enter the conference.
212 * Added a new option for conference recording, record_file_append. If enabled,
213 when the recording is stopped and then re-started, the existing recording
214 will be used and appended to.
216 * ConfBridge now has the ability to set the language of announcements to the
217 conference. The language can be set on a bridge profile in confbridge.conf
218 or by the dialplan function CONFBRIDGE(bridge,language)=en.
222 * The channel variable CPLAYBACKSTATUS may now return the value
223 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
224 such as AMI. See the AMI action ControlPlayback for more information.
228 * Added the 'a' option, which allows the caller to enter in an additional
229 alias for the user in the directory. This option must be used in conjunction
230 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
231 specified in voicemail.conf.
235 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
236 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
237 containing the unique ID of the bridge that the channel happens to be in.
241 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
242 for more information.
244 * Variables are no longer purged from the original CDR. See the 'v' option for
247 * The 'A' option has been removed. The Answer time on a CDR is never updated
250 * The 'd' option has been removed. The disposition on a CDR is a function of
251 the state of the channel and cannot be altered.
253 * The 'D' option has been removed. Who the Party B is on a CDR is a function
254 of the state of the respective channels involved in the CDR and cannot be
257 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
258 such that the start time and, if applicable, the answer time was updated.
259 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
260 'r' option now triggers the Reset, setting the start time (and answer time
261 if applicable) to the current time. Note that the 'a' option still sets
262 the answer time to the current time if the channel was already answered.
264 * The 's' option has been removed. A variable can be set on the original CDR
265 if desired using the CDR function, and removed from a forked CDR using the
268 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
269 longer applies in the CDR engine.
271 * The 'v' option now prevents the copy of the variables from the original CDR
272 to the forked CDR. Previously the variables were always copied but were
273 removed from the original. This was changed as removing variables from a CDR
274 can have unintended side effects - this option allows the user to prevent
275 propagation of variables from the original to the forked without modifying
280 * Added the 'n' option to MeetMe to prevent application of the DENOISE
281 function to a channel joining a conference. Some channel drivers that vary
282 the number of audio samples in a voice frame will experience significant
283 quality problems if a denoiser is attached to the channel; this option gives
284 them the ability to remove the denoiser without having to unload func_speex.
288 * The 'b' option now includes conferences as well as sounds played to the
291 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
292 running during a transfer. If a MixMonitor is started on a channel,
293 the MixMonitor will continue to record the audio passing through the
294 channel even in the presence of transfers.
298 * The NoCDR application is deprecated. Please use the CDR_PROP function to
301 * While the NoCDR application will prevent CDRs for a channel from being
302 propagated to registered CDR backends, it will not prevent that data from
303 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
304 function that enables CDRs on a channel will restore those records that have
305 not yet been finalized.
309 * The app_parkandannounce module has been removed. The application
310 ParkAndAnnounce is now provided by the res_parking module. See the
311 res_parking changes for more information.
315 * Added queue available hint. The hint can be added to the dialplan using the
316 following syntax: exten,hint,Queue:{queue_name}_avail
317 For example, if the name of the queue is 'markq':
318 exten => 8501,hint,Queue:markq_avail
319 This will report 'InUse' if there are no logged in agents or no free agents.
320 It will report 'Idle' when an agent is free.
322 * Queues now support a hint for member paused state. The hint uses the form
323 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
324 are the name of the queue and the name of the member to subscribe to,
325 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
326 Members will show as In Use when paused.
328 * The configuration options eventwhencalled and eventmemberstatus have been
329 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
330 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
331 sent. The "Variable" fields will also no longer exist on the Agent* events.
332 These events can be filtered out from a connected AMI client using the
333 eventfilter setting in manager.conf.
335 * The queue log now differentiates between blind and attended transfers. A
336 blind transfer will result in a BLINDTRANSFER message with the destination
337 context and extension. An attended transfer will result in an
338 ATTENDEDTRANSFER message. This message will indicate the method by which
339 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
340 for running an application on a bridge or channel, or "LINK" for linking
341 two bridges together with local channels. The queue log will also now detect
342 externally initiated blind and attended transfers and record the transfer
345 * When performing queue pause/unpause on an interface without specifying an
346 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
347 least one member of any queue exists for that interface.
349 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
350 for realtime queue log entries.
354 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
355 CDRs when they were previously disabled on a channel.
357 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
358 backends occurs on an as-needed basis in order to preserve linkedid
359 propagation and other needed behavior.
363 * A new application, this is similar to SayAlpha except that it supports
364 case sensitive playback of the specified characters. For example,
365 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
369 * This application is deprecated in favor of CHANNEL(amaflags).
373 * The SendDTMF application will now accept 'W' as valid input. This will cause
374 the application to delay one second while streaming DTMF.
378 * A new application in Asterisk 12, this hands control of the channel calling
379 the application over to an external system. Currently, external systems
380 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
384 * UserEvent will now handle duplicate keys by overwriting the previous value
387 * In addition to AMI, UserEvent invocations will now be distributed to any
388 interested Stasis applications.
392 * The voicemail.conf configuration file now has an 'alias' configuration
393 parameter for use with the Directory application. The voicemail realtime
394 database table schema has also been updated with an 'alias' column.
399 * Pass through support has been added for both VP8 and Opus.
401 * Added format attribute negotiation for the Opus codec. Format attribute
402 negotiation is provided by the res_format_attr_opus module.
407 * Masquerades as an operation inside Asterisk have been effectively hidden
408 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
409 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
410 dropping of frame/audio hooks, and other internal implementation details
411 that users had to deal with. This fundamental change has large implications
412 throughout the changes documented for this version. For more information
413 about the new core architecture of Asterisk, please see the Asterisk wiki.
415 * Multiple parties in a bridge may now be transferred. If a participant in a
416 multi-party bridge initiates a blind transfer, a Local channel will be used
417 to execute the dialplan location that the transferer sent the parties to. If
418 a participant in a multi-party bridge initiates an attended transfer,
419 several options are possible. If the attended transfer results in a transfer
420 to an application, a Local channel is used. If the attended transfer results
421 in a transfer to another channel, the resulting channels will be merged into
424 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
425 driver specific. If the channel variable is set on the transferrer channel,
426 the sound will be played to the target of an attended transfer.
428 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
429 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
430 listed. Any more peers in the bridge will not be included in the list.
431 BRIDGEPEER is not valid in holding bridges like parking since those channels
432 do not talk to each other even though they are in a bridge.
434 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
435 and will contain a value if the BRIDGEPEER's channel driver supports it.
437 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
438 was responsible for an attended transfer in a similar fashion to
441 * Modules using the Configuration Framework or Sorcery must have XML
442 configuration documentation. This configuration documentation is included
443 with the rest of Asterisk's XML documentation, and is accessible via CLI
444 commands. See the CLI changes for more information.
446 AMI (Asterisk Manager Interface)
448 * Major changes were made to both the syntax as well as the semantics of the
449 AMI protocol. In particular, AMI events have been substantially improved
450 in this version of Asterisk. For more information, please see the AMI
451 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
453 * AMI events that reference a particular channel or bridge will now always
454 contain a standard set of fields. When multiple channels or bridges are
455 referenced in an event, fields for at least some subset of the channels
456 and bridges in the event will be prefixed with a descriptive name to avoid
457 name collisions. See the AMI event documentation on the Asterisk wiki for
460 * The CLI command 'manager show commands' no longer truncates command names
461 longer than 15 characters and no longer shows authorization requirement
462 for commands. 'manager show command' now displays the privileges needed
463 for using a given manager command instead.
465 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
466 peer in its response if the peer has a subscribe context set.
468 * The SIPqualifypeer action now acknowledges the request once it has
469 established that the request is against a known peer. It also issues a new
470 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
472 * The PlayDTMF action now supports an optional 'Duration' parameter. This
473 specifies the duration of the digit to be played, in milliseconds.
475 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
476 updates when changes occur instead of requiring the use of pollmailboxes.
478 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
479 AMI client to manipulate audio currently being played back on a channel. The
480 supported operations depend on the application being used to send audio to
481 the channel. When the audio playback was initiated using the ControlPlayback
482 application or CONTROL STREAM FILE AGI command, the audio can be paused,
483 stopped, restarted, reversed, or skipped forward. When initiated by other
484 mechanisms (such as the Playback application), the audio can be stopped,
485 reversed, or skipped forward.
487 * Channel related events now contain a snapshot of channel state, adding new
488 fields to many of these events.
490 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
491 in a future release. Please use the common 'Exten' field instead.
493 * The AMI event 'UserEvent' from app_userevent now contains the channel state
494 fields. The channel state fields will come before the body fields.
496 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
497 'UnParkedCall' have changed significantly in the new res_parking module.
499 The 'Channel' and 'From' headers are gone. For the channel that was parked
500 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
501 has a number of fields associated with it. The old 'Channel' header relayed
502 the same data as the new 'ParkeeChannel' header.
504 The 'From' field was ambiguous and changed meaning depending on the event.
505 for most of these, it was the name of the channel that parked the call
506 (the 'Parker'). There is no longer a header that provides this channel name,
507 however the 'ParkerDialString' will contain a dialstring to redial the
508 device that parked the call.
510 On UnParkedCall events, the 'From' header would instead represent the
511 channel responsible for retrieving the parkee. It receives a channel
512 snapshot labeled 'Retriever'. The 'from' field is is replaced with
515 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
517 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
518 fashion has changed the field names 'StartExten' and 'StopExten' to
519 'StartSpace' and 'StopSpace' respectively.
521 * The deprecated use of | (pipe) as a separator in the channelvars setting in
522 manager.conf has been removed.
524 * Channel Variables conveyed with a channel no longer contain the name of the
525 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
526 ChanVariable: bar=baz. When multiple channels are present in a single AMI
527 event, the various ChanVariable fields will contain a suffix that specifies
528 which channel they correspond to.
530 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
531 event always conveys the AMI event for a particular channel.
533 * All 'Reload' events have been consolidated into a single event type. This
534 event will always contain a Module field specifying the name of the module
535 and a Status field denoting the result of the reload. All modules now issue
536 this event when being reloaded.
538 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
539 fail to receive this event due to being connected after modules have loaded.
540 AMI connections that want to know when Asterisk is ready should listen for
541 the 'FullyBooted' event.
543 * app_fax now sends the same send fax/receive fax events as res_fax. The
544 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
545 now the 'ReceiveFAX' event.
547 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
548 'MusicOnHoldStop'. The sub type field has been removed.
550 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
551 carrier for another protocol.
553 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
554 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
555 to the specific channel. 'Both' may be specified to play a tone to both
556 channels. The old 'yes' option is still accepted as a way of playing the
557 tone to Channel2 only.
559 * The AMI 'Status' response event to the AMI Status action replaces the
560 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
561 indicate what bridge the channel is currently in.
563 * The AMI 'Hold' event has been moved out of individual channel drivers, into
564 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
567 * The AMI events in app_queue have been made more consistent with each other.
568 Events that reference channels (QueueCaller* and Agent*) will show
569 information about each channel. The (infamous) 'Join' and 'Leave' AMI
570 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
572 * The 'MCID' AMI event now publishes a channel snapshot when available and
573 its non-channel-snapshot parameters now use either the "MCallerID" or
574 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
575 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
576 parameters in the channel snapshot.
578 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
579 'AgentLogin' and 'AgentLogoff' respectively.
581 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
582 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
584 * 'ChannelUpdate' events have been removed.
586 * All AMI events now contain a 'SystemName' field, if available.
588 * Local channel optimization is now conveyed in two events:
589 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
590 when the Local channel driver begins attempting to optimize itself out of
591 the media path; the End event is sent after the channel halves have
592 successfully optimized themselves out of the media path.
594 * Local channel information in events is now prefixed with 'LocalOne' and
595 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
596 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
597 and 'LocalOptimizationEnd' events.
599 * The option 'allowmultiplelogin' can now be set or overriden in a particular
600 account. When set in the general context, it will act as the default
601 setting for defined accounts.
603 * The 'BridgeAction' event was removed. It technically added no value, as the
604 Bridge Action already receives confirmation of the bridge through a
605 successful completion Event.
607 * The 'BridgeExec' events were removed. These events duplicated the events that
608 occur in the Briding API, and are conveyed now through BridgeCreate,
609 BridgeEnter, and BridgeLeave events.
611 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
612 previous versions. They now report all SR/RR packets sent/received, and
613 have been restructured to better reflect the data sent in a SR/RR. In
614 particular, the event structure now supports multiple report blocks.
616 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
617 raised when a blind transfer/attended transfer completes successfully.
618 They contain information about the transfer that just completed, including
619 the location of the transfered channel.
621 * Added a 'security' class to AMI which outputs the required fields for
622 security messages similar to the log messages from res_security_log
624 CDR (Call Detail Records)
626 * Significant changes have been made to the behavior of CDRs. The CDR engine
627 was effectively rewritten and built on the Stasis message bus. For a full
628 definition of CDR behavior in Asterisk 12, please read the specification
629 on the Asterisk wiki (wiki.asterisk.org).
631 * CDRs will now be created between all participants in a bridge. For each
632 pair of channels in a bridge, a CDR is created to represent the path of
633 communication between those two endpoints. This lets an end user choose who
634 to bill for what during bridge operations with multiple parties.
636 * The duration, billsec, start, answer, and end times now reflect the times
637 associated with the current CDR for the channel, as opposed to a cumulative
638 measurement of all CDRs for that channel.
640 * When a CDR is dispatched, user defined CDR variables from both parties are
641 included in the resulting CDR. If both parties have the same variable, only
642 the Party A value is provided.
644 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
645 information regarding the CDR engine is logged as verbose messages. This
646 option should only be used if the behavior of the CDR engine needs to be
649 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
650 normally configured in cdr.conf.
652 * Added CLI command 'cdr show active {channel}'. When {channel} is not
653 specified, this command provides a summary of the channels with CDR
654 information and their statistics. When {channel} is specified, it shows
655 detailed information about all records associated with {channel}.
657 CEL (Channel Event Logging)
659 * CEL has undergone significant rework in Asterisk 12, and is now built on the
660 Stasis message bus. Please see the specification for CEL on the Asterisk
661 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
664 * The 'extra' field of all CEL events that use it now consists of a JSON blob
665 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
667 * BLINDTRANSFER events now report the transferee bridge unique
668 identifier, extension, and context in a JSON blob as the extra string
669 instead of the transferee channel name as the peer.
671 * ATTENDEDTRANSFER events now report the peer as NULL and additional
672 information in the 'extra' string as a JSON blob. For transfers that occur
673 between two bridged channels, the 'extra' JSON blob contains the primary
674 bridge unique identifier, the secondary channel name, and the secondary
675 bridge unique identifier. For transfers that occur between a bridged channel
676 and a channel running an app, the 'extra' JSON blob contains the primary
677 bridge unique identifier, the secondary channel name, and the app name.
679 * LOCAL_OPTIMIZE events have been added to convey local channel
680 optimizations with the record occurring for the semi-one channel and
681 the semi-two channel name in the peer field.
683 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
684 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
685 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
686 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
687 regardless of whether or not that bridge happens to contain multiple
692 * When compiled with '--enable-dev-mode', the astobj2 library will now add
693 several CLI commands that allow for inspection of ao2 containers that
694 register themselves with astobj2. The CLI commands are 'astobj2 container
695 dump', 'astobj2 container stats', and 'astobj2 container check'.
697 * Added specific CLI commands for bridge inspection. This includes 'bridge
698 show all', which lists all bridges in the system, and 'bridge show {id}',
699 which provides specific information about a bridge.
701 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
702 ejecting the channels currently in the bridge. If the channels cannot
703 continue in the dialplan or application that put them in the bridge, they
706 * Added command 'bridge kick'. This will eject a single channel from a bridge.
708 * Added commands to inspect and manipulate the registered bridge technologies.
709 This include 'bridge technology show', which lists the registered bridge
710 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
711 which controls whether or not a registered bridge technology can be used
712 during smart bridge operations. If a technology is suspended, it will not
713 be used when a bridge technology is picked for channels; when unsuspended,
714 it can be used again.
716 * The command 'config show help {module} {type} {option}' will show
717 configuration documentation for modules with XML configuration
718 documentation. When {module}, {type}, and {option} are omitted, a listing
719 of all modules with registered documentation is displayed. When {module}
720 is specified, a listing of all configuration types for that module is
721 displayed, along with their synopsis. When {module} and {type} are
722 specified, a listing of all configuration options for that type are
723 displayed along with their synopsis. When {module}, {type}, and {option}
724 are specified, detailed information for that configuration option is
727 * Added 'core show sounds' and 'core show sound' CLI commands. These display
728 a listing of all installed media sounds available on the system and
729 detailed information about a sound, respectively.
731 * 'xmldoc dump' has been added. This CLI command will dump the XML
732 documentation DOM as a string to the specified file. The Asterisk core
733 will populate certain XML elements pulled from the source files with
734 additional run-time information; this command lets a user produce the
735 XML documentation with all information.
739 * Parking has been pulled from core and placed into a separate module called
740 res_parking. See Parking changes below for more details. Configuration for
741 parking should now be performed in res_parking.conf. Configuration for
742 parking in features.conf is now unsupported.
744 * Core attended transfers now have several new options. While performing an
745 attended transfer, the transferer now has the following options:
746 - *1 - cancel the attended transfer (configurable via atxferabort)
747 - *2 - complete the attended transfer, dropping out of the call
748 (configurable via atxfercomplete)
749 - *3 - complete the attended transfer, but stay in the call. This will turn
750 the call into a multi-party bridge (configurable via atxferthreeway)
751 - *4 - swap to the other party. Once an attended transfer has begun, this
752 options may be used multiple times (configurable via atxferswap)
754 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
755 must be on the channel initiating the transfer to have any effect.
757 * The BRIDGE_FEATURES channel variable would previously only set features for
758 the calling party and would set this feature regardless of whether the
759 feature was in caps or in lowercase. Use of a caps feature for a letter
760 will now apply the feature to the calling party while use of a lowercase
761 letter will apply that feature to the called party.
763 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
765 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
766 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
767 activated the dynamic feature.
769 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
770 only on the channel executing the dynamic feature. Executing a dynamic
771 feature on the bridge peer in a multi-party bridge will execute it on all
772 peers of the activating channel.
774 * You can now have the settings for a channel updated using the FEATURE()
775 and FEATUREMAP() functions inherited to child channels by setting
776 FEATURE(inherit)=yes.
778 * automixmon now supports additional channel variables from automon including:
779 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
780 and TOUCH_MIXMONITOR_MESSAGE_STOP
782 * A new general features.conf option 'recordingfailsound' has been added which
783 allowssetting a failure sound for a user tries to invoke a recording feature
784 such as automon or automixmon and it fails.
786 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
787 features.c for atxferdropcall=no to work properly. This option now just
792 * Added log rotation strategy 'none'. If set, no log rotation strategy will
793 be used. Given that this can cause the Asterisk log files to grow quickly,
794 this option should only be used if an external mechanism for log management
799 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
800 will store the path information for that peer when it registers. Realtime
801 tables can also use the 'supportpath' field to enable Path header support.
803 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
804 objectIdentifier. This maps to the supportpath option in sip.conf.
808 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
809 provides modules a useful abstraction on top of the many storage mechanisms
810 in Asterisk, including the Asterisk Database, static configuration files,
811 static Realtime, and dynamic Realtime. It also provides a caching service.
812 Users can configure a hierarchy of data storage layers for specific modules
815 * All future modules which utilize Sorcery for object persistence must have a
816 column named "id" within their schema when using the Sorcery realtime module.
817 This column must be able to contain a string of up to 128 characters in length.
819 Security Events Framework
821 * Security Event timestamps now use ISO 8601 formatted date/time instead of
822 the "seconds-microseconds" format that it was using previously.
826 * The Stasis message bus is a publish/subscribe message bus internal to
827 Asterisk. Many services in Asterisk are built on the Stasis message bus,
828 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
829 Stasis can be configured in stasis.conf. Note that these parameters operate
830 at a very low level in Asterisk, and generally will not require changes.
834 * When a channel driver is configured to enable jiterbuffers, they are now
835 applied unconditionally when a channel joins a bridge. If a jitterbuffer
836 is already set for that channel when it enters, such as by the JITTERBUFFER
837 function, then the existing jitterbuffer will be used and the one set by
838 the channel driver will not be applied.
842 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
843 dialplan applications provided by the app_agent_pool module. Agents are
844 connected with callers using the new AgentRequest dialplan application.
845 The Agents:<agent-id> device state is available to monitor the status of an
846 agent. See agents.conf.sample for valid configuration options.
848 * The updatecdr option has been removed. Altering the names of channels on a
849 CDR is not supported - the name of the channel is the name of the channel,
850 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
851 has also been removed, for the same reason.
853 * The endcall and enddtmf configuration options are removed. Use the
854 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
855 channel before calling AgentLogin.
859 * chan_bridge has been removed. Its functionality has been incorporated
860 directly into the ConfBridge application itself.
864 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
865 of the specified span and its B-channels. Note that this command should
866 only be used if you understand the risks it entails.
868 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
869 A range of channels can be specified to be destroyed. Note that this command
870 should only be used if you understand the risks it entails.
872 * Added the CLI command 'dahdi create channels'. A range of channels can be
873 specified to be created, or the keyword 'new' can be used to add channels
878 * IPv6 support has been added. We are now able to bind to and
879 communicate using IPv6 addresses.
883 * The /b option has been removed.
885 * chan_local moved into the system core and is no longer a loadable module.
889 * Added general support for busy detection.
891 * Added ECAM command support for Sony Ericsson phones.
895 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
896 SIP stack. A collection of resource modules provides the bulk of the SIP
897 functionality. For more information on the new SIP channel driver, see
898 https://wiki.asterisk.org/wiki/x/JYGLAQ
902 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
903 using the 'supportpath' setting, either on a global basis or on a peer basis.
904 This setting enables Asterisk to route outgoing out-of-dialog requests via a
905 set of proxies by using a pre-loaded route-set defined by the Path headers in
906 the REGISTER request. See Realtime updates for more configuration information.
908 * The SIP_CODEC family of variables may now specify more than one codec. Each
909 codec must be separated by a comma. The first codec specified is the
910 preferred codec for the offer. This allows a dialplan writer to specify both
911 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
913 * The 'callevents' parameter has been removed. Hold AMI events are now raised
914 in the core, and can be filtered out using the 'eventfilter' parameter
917 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
918 codecs configured for a peer instead of the requested codec.
920 * The option "register_retry_403" has been added to chan_sip to work around
921 servers that are known to erroneously send 403 in response to valid
922 REGISTER requests and allows Asterisk to continue attepmting to connect.
926 * Added the 'immeddialkey' parameter. If set, when the user presses the
927 configured key the already entered number will be immediately dialed. This
928 is useful when the dialplan allows for variable length pattern matching.
929 Valid options are '*' and '#'.
931 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
932 milliseconds) before a call forward is considered to not be answered.
934 * The 'serviceurl' parameter allows Service URLs to be attached to line
943 * The password option has been disabled, as the AgentLogin application no
944 longer provides authentication.
948 * Due to changes in the Asterisk core, this function is no longer needed to
949 preserve a MixMonitor on a channel during transfer operations and dialplan
950 execution. It is effectively obsolete.
954 * The 'amaflags' and 'accountcode' attributes for the CDR function are
955 deprecated. Use the CHANNEL function instead to access these attributes.
957 * The 'l' option has been removed. When reading a CDR attribute, the most
958 recent record is always used. When writing a CDR attribute, all non-finalized
961 * The 'r' option has been removed, for the same reason as the 'l' option.
963 * The 's' option has been removed, as LOCKED semantics no longer exist in the
968 * A new function CDR_PROP has been added. This function lets you set properties
969 on a channel's active CDRs. This function is write-only. Properties accept
970 boolean values to set/clear them on the channel's CDRs. Valid properties
972 - 'party_a' - make this channel the preferred Party A in any CDR between two
973 channels. If two channels have this property set, the creation time of the
974 channel is used to determine who is Party A. Note that dialed channels are
975 never Party A in a CDR.
976 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
977 application when set to True, and analogous to the 'e' option in ResetCDR
982 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
983 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
984 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
987 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
988 string, i.e., [[context],extension],priority. If set on a channel, if a
989 channel leaves a bridge but is not hung up it will resume dialplan execution
994 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
995 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
996 The value of this setting is ignored when disabled is used for the argument.
1000 * A new function provided by chan_pjsip, this function can be used in
1001 conjunction with the Dial application to construct a dial string that will
1002 dial all contacts on an Address of Record associated with a chan_pjsip
1007 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1008 outbound channel prior to dialing.
1012 * Redirecting reasons can now be set to arbitrary strings. This means
1013 that the REDIRECTING dialplan function can be used to set the redirecting
1014 reason to any string. It also allows for custom strings to be read as the
1015 redirecting reason from SIP Diversion headers.
1019 * The SPEECH_ENGINE function now supports read operations. When read from, it
1020 will return the current value of the requested attribute.
1026 res_agi (Asterisk Gateway Interface)
1028 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1030 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1033 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1034 will start the playback of the audio at the position specified. It will
1035 also return the final position of the file in 'endpos'.
1037 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1038 channel variable if the user stopped the file playback or if a remote
1039 entity stopped the playback. If neither stopped the playback, it will
1040 indicate the overall success/failure of the playback. If stopped early,
1041 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1044 * The SAY ALPHA command now accepts an additional parameter to control
1045 whether it specifies the case of uppercase, lowercase, or all letters to
1046 provide functionality similar to SayAlphaCase.
1048 res_ari (Asterisk RESTful Interface) (and others)
1050 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1051 control telephony primitives in Asterisk by remote client. This includes
1052 channels, bridges, endpoints, media, and other fundamental concepts. Users
1053 of ARI can develop their own communications applications, controlling
1054 multiple channels using an HTTP RESTful interface and receiving JSON events
1055 about the objects via a WebSocket connection. ARI can be configured in
1056 Asterisk via ari.conf. For more information on ARI, see
1057 https://wiki.asterisk.org/wiki/x/0YCLAQ
1061 * Parking has been extracted from the Asterisk core as a loadable module,
1062 res_parking. Configuration for parking is now provided by res_parking.conf.
1063 Configuration through features.conf is no longer supported.
1065 * res_parking uses the configuration framework. If an invalid configuration is
1066 supplied, res_parking will fail to load or fail to reload. Previously,
1067 invalid configurations would generally be accepted, with certain errors
1068 resulting in individually disabled parking lots.
1070 * Parked calls are now placed in bridges. While this is largely an
1071 architectural change, it does have implications on how channels in a parking
1072 lot are viewed. For example, commands that display channels in bridges will
1073 now also display the channels in a parking lot.
1075 * The order of arguments for the new parking applications have been modified.
1076 Timeout and return context/exten/priority are now implemented as options,
1077 while the name of the parking lot is now the first parameter. See the
1078 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1079 in-depth information as well as syntax.
1081 * Extensions are by default no longer automatically created in the dialplan to
1082 park calls or pickup parked calls. Generation of dialplan extensions can be
1083 enabled using the 'parkext' configuration option.
1085 * ADSI functionality for parking is no longer supported. The 'adsipark'
1086 configuration option has been removed as a result.
1088 * The PARKINGSLOT channel variable has been deprecated in favor of
1089 PARKING_SPACE to match the naming scheme of the new system.
1091 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1092 channel even when the configuration option 'comebactoorigin' is enabled.
1094 * A new CLI command 'parking show' has been added. This allows a user to
1095 inspect the parking lots that are currently in use.
1096 'parking show <parkinglot>' will also show the parked calls in a specific
1099 * The CLI command 'parkedcalls' is now deprecated in favor of
1100 'parking show <parkinglot>'.
1102 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1103 can be used to get a list of parked calls for a specific parking lot.
1105 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1106 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1107 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1108 longer a required argument.
1110 * The ParkAndAnnounce application is now provided through res_parking instead
1111 of through the separate app_parkandannounce module.
1113 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1114 by default. Instead, it will follow the timeout rules of the parking lot. The
1115 old behavior can be reproduced by using the 'c' option.
1117 * Dynamic parking lots will now fail to be created under the following
1119 - if the parking lot specified by PARKINGDYNAMIC does not exist
1120 - if they require exclusive park and parkedcall extensions which overlap
1121 with existing parking lots.
1123 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1124 currently contain no calls. Dynamic parking lots containing parked calls
1125 will persist through the reloads without alteration.
1127 * If 'parkext_exclusive' is set for a parking lot and that extension is
1128 already in use when that parking lot tries to register it, this is now
1129 considered a parking system configuration error. Configurations which do
1130 this will be rejected.
1132 * Added channel variable PARKER_FLAT. This contains the name of the extension
1133 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1134 comebacktoorigin is disabled, but the dialplan or an external control
1135 mechanism wants to use the extension in the park-dial context that was
1136 generated to re-dial the parker on timeout.
1138 res_pjsip (and many others)
1140 * A large number of resource modules make up the SIP stack based on pjsip.
1141 The chan_pjsip channel driver users these resource modules to provide
1142 various SIP functionality in Asterisk. The majority of configuration for
1143 these modules is performed in pjsip.conf. Other modules may use their
1144 own configuration files.
1148 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1149 them, an Asterisk-specific version of PJSIP needs to be installed.
1150 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1152 res_statsd/res_chan_stats
1154 * A new resource module, res_statsd, has been added, which acts as a statsd
1155 client. This module allows Asterisk to publish statistics to a statsd
1156 server. In conjunction with res_chan_stats, it will publish statistics about
1157 channels to the statsd server. It can be configured via res_statsd.conf.
1161 * Device state for XMPP buddies is now available using the following format:
1162 XMPP/<client name>/<buddy address>
1163 If any resource is available the device state is considered to be not in use.
1164 If no resources exist or all are unavailable the device state is considered
1171 Realtime/Database Scripts
1173 * Asterisk previously included example db schemas in the contrib/realtime/
1174 directory of the source tree. This has been replaced by a set of database
1175 migrations using the Alembic framework. This allows you to use alembic to
1176 initialize the database for you. It will also serve as a database migration
1177 tool when upgrading Asterisk in the future.
1179 See contrib/ast-db-manage/README.md for more details.
1183 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1184 This python script will convert an existing sip.conf file to a
1185 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1186 is meant to be an aid in converting an existing chan_sip configuration to
1187 a chan_pjsip configuration, but it is expected that configuration beyond
1188 what the script provides will be needed.
1191 ------------------------------------------------------------------------------
1192 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1193 ------------------------------------------------------------------------------
1197 * The Asterisk build system will now build and install a shared library
1198 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1199 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1200 that Asterisk can ensure that these functions do *not* get called by any
1201 modules that are loaded into Asterisk, since they should only be called once
1202 in any single process. If desired, this feature can be disabled by supplying
1203 the "--disable-asteriskssl" option to the configure script.
1205 * A new make target, 'full', has been added to the Makefile. This performs
1206 the same compilation actions as make all, but will also scan the entirety of
1207 each source file for documentation. This option is needed to generate AMI
1208 event documentation. Note that your system must have Python in order for
1209 this make target to succeed.
1211 * The optimization portion of the build system has been reworked to avoid
1212 broken builds on certain architectures. All architecture-specific
1213 optimization has been removed in favor of using -march=native to allow gcc
1214 to detect the environment in which it is running when possible. This can
1215 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1217 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1218 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1220 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1221 previously parsed the header file to obtain the version of Asterisk, you
1222 will now have to go through Asterisk to get the version information.
1230 * Added 'F()' option. Similar to the dial option, this can be supplied with
1231 arguments indicating where the callee should go after the caller is hung up,
1232 or without options specified, the priority after the Queue will be used.
1237 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1238 all non-admin participants on a conference. The confbridge configuration
1239 file also allows for the default sounds played to all conference users when
1240 this occurs to be overriden using sound_participants_unmuted and
1241 sound_participants_muted.
1243 * Added menu action participant_count. This will playback the number of
1244 current participants in a conference.
1246 * Added announcement configuration option to user profile. If set the sound
1247 file will be played to the user, and only the user, upon joining the
1250 * Added record_file_append option that defaults to "yes", but if set to no
1251 will create a new file between each start/stop recording.
1256 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1257 channels respectively before the callee channels are called.
1262 * Added support for IPv6.
1264 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1265 external process will cause the current playlist to be cleared, including
1266 stopping any audio file that is currently playing. This is useful when you
1267 want to interrupt audio playback only when specific DTMF is entered by the
1273 * A new option, 'I' has been added to app_followme. By setting this option,
1274 Asterisk will not update the caller with connected line changes when they
1275 occur. This is similar to app_dial and app_queue.
1277 * The 'N' option is now ignored if the call is already answered.
1279 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1280 and caller channels respectively before the callee channels are called.
1282 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1288 * MixMonitor hooks now have IDs associated with them which can be used to
1289 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1290 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1291 now accepts that ID as an argument.
1293 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1294 indicated mailboxes.
1299 * The connect action in app_mysql now allows you to specify a port number to
1300 connect to. This is useful if you run a MySQL server on a non-standard
1306 * Increased the default number of allowed destinations from 5 to 12.
1311 * The app_page application now no longer depends on DAHDI or app_meetme. It
1312 has been re-architected to use app_confbridge internally.
1317 * Added queue options autopausebusy and autopauseunavail for automatically
1318 pausing a queue member when their device reports busy or congestion.
1320 * The 'ignorebusy' option for queue members has been deprecated in favor of
1321 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1322 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1323 per interface basis. Individual ringinuse values can now be set in
1324 queues.conf via an argument to member definitions. Lastly, the queue
1325 'ringinuse' setting now only determines defaults for the per member
1326 'ringinuse' setting and does not override per member settings like it does
1327 in earlier versions.
1329 * Added 'F()' option. Similar to the dial option, this can be supplied with
1330 arguments indicating where the callee should go after the caller is hung up,
1331 or without options specified, the priority after the Queue will be used.
1333 * Added new option log_member_name_as_agent, which will cause the membername to
1334 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1335 state_interface has been set.
1337 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1339 * App_queue will now play periodic announcements for the caller that
1340 holds the first position in the queue while waiting for answer.
1344 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1345 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1346 changed arguments to SayUnixTime so that every option is truly optional even
1347 when using multiple options (so that j option could be used without having to
1348 manually specify timezone and format) There are other benefits, e.g., format
1349 can now be used without specifying time zone as well.
1354 * Addition of the VM_INFO function - see Function changes.
1356 * The imapserver, imapport, and imapflags configuration options can now be
1357 overriden on a user by user basis.
1359 * When voicemail plays a message's envelope with saycid set to yes, when
1360 reaching the caller id field it will play a recording of a file with the same
1361 base name as the sender's callerid if there is a similarly named file in
1362 <astspooldir>/recordings/callerids/
1364 * Voicemails now contains a unique message identifier "msg_id", which is stored
1365 in the message envelope with the sound files. IMAP backends will now store
1366 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1367 backends will store the message identifier in a "msg_id" column. See
1368 UPGRADE.txt for more information.
1370 * Added VoiceMailPlayMsg application. This application will play a single
1371 voicemail message from a mailbox. The result of the application, SUCCESS or
1372 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1377 * Hangup handlers can be attached to channels using the CHANNEL() function.
1378 Hangup handlers will run when the channel is hung up similar to the h
1379 extension. The hangup_handler_push option will push a GoSub compatible
1380 location in the dialplan onto the channel's hangup handler stack. The
1381 hangup_handler_pop option will remove the last added location, and optionally
1382 replace it with a new GoSub compatible location. The hangup_handler_wipe
1383 option will remove all locations on the stack, and optionally add a new
1386 * The expression parser now recognizes the ABS() absolute value function,
1387 which will convert negative floating point values to positive values.
1389 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1390 control of faxdetect.
1392 * Addition of the VM_INFO function that can be used to retrieve voicemail
1393 user information, such as the email address and full name.
1394 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1397 * The REDIRECTING function now supports the redirecting original party id
1400 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1401 lets you set some of the configuration options from the [general] section
1402 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1403 the key sequence used to activate built-in features, such as blindxfer,
1404 and automon. See the built-in documentation for details.
1406 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1407 instead of simply the uri. This is the format that MessageSend() can use
1408 in the from parameter for outgoing SIP messages.
1410 * Added the PRESENCE_STATE function. This allows retrieving presence state
1411 information from any presence state provider. It also allows setting
1412 presence state information from a CustomPresence presence state provider.
1413 See AMI/CLI changes for related commands.
1415 * Added the AMI_CLIENT function to make manager account attributes available
1416 to the dialplan. It currently supports returning the current number of
1417 active sessions for a given account.
1419 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1420 and the REDIRECTING functions.
1428 * Added a manager event "LocalBridge" for local channel call bridges between
1429 the two pseudo-channels created.
1434 * Added dialtone_detect option for analog ports to disconnect incoming
1435 calls when dialtone is detected.
1437 * Added option colp_send to send ISDN connected line information. Allowed
1438 settings are block, to not send any connected line information; connect, to
1439 send connected line information on initial connect; and update, to send
1440 information on any update during a call. Default is update.
1442 * Add options namedcallgroup and namedpickupgroup to support installations
1443 where a higher number of groups (>64) is required.
1445 * Added support to use private party ID information with PRI calls.
1450 * A new channel driver named chan_motif has been added which provides support for
1451 Google Talk and Jingle in a single channel driver. This new channel driver includes
1452 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1453 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1454 specification, current Google Jingle specification, and the original Google Talk
1460 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1461 globally and overriden on a peer by peer basis.
1463 * Direct media functionality has been added. Options in config are:
1464 directmedia (directrtp) and directrtpsetup (earlydirect)
1466 * ChannelUpdate events now contain a CallRef header.
1471 * Asterisk will no longer substitute CID number for CID name in the display
1472 name field if CID number exists without a CID name. This change improves
1473 compatibility with certain device features such as Avaya IP500's directory
1476 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1477 created using that setting to not be removed during SIP reload.
1479 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1480 request with a "Record:" header, this will turn the requested feature on/off.
1481 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1482 dynamic features must be enabled and configured properly on the requesting
1483 channel for this to function properly.
1485 * Add support to realtime for the 'callbackextension' option.
1487 * When multiple peers exist with the same address, but differing
1488 callbackextension options, incoming requests that are matched by address
1489 will be matched to the peer with the matching callbackextension if it is
1492 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1493 which set the force_rport and comedia options automatically if Asterisk
1494 detects that an incoming SIP request crossed a NAT after being sent by
1495 the remote endpoint.
1497 * The default global nat setting in sip.conf has been changed from force_rport
1498 to auto_force_rport.
1500 * NAT settings are now a combinable list of options. The equivalent of the
1501 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1503 * Adds an option send_diversion which can be disabled to prevent
1504 diversion headers from automatically being added to INVITE requests.
1506 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1507 be sent to the remote host at a given interval to keep the NAT mapping open.
1508 This can be enabled using the keepalive configuration option.
1510 * Add option 'tonezone' to specify country code for indications. This option
1511 can be set both globally and overridden for specific peers.
1513 * The SIP Security Events Framework now supports IPv6.
1515 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1516 between multiple user agents. When set, for directmedia reinvites,
1517 Asterisk will not send an immediate reinvite on an incoming call leg. This
1518 option is useful when peered with another SIP user agent that is known to
1519 send immediate direct media reinvites upon call establishment.
1521 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1524 * Add options subminexpiry and submaxexpiry to set limits of subscription
1525 timer independently from registration timer settings. The setting of the
1526 registration timer limits still is done by options minexpiry, maxexpiry
1527 and defaultexpiry. For backwards compatibility the setting of minexpiry
1528 and maxexpiry also is used to configure the subscription timer limits if
1529 subminexpiry and submaxexpiry are not set in sip.conf.
1531 * Set registration timer limits to default values when reloading sip
1532 configuration and values are not set by configuration.
1534 * Add options namedcallgroup and namedpickupgroup to support installations
1535 where a higher number of groups (>64) is required.
1537 * When a MESSAGE request is received, the address the request was received from
1538 is now saved in the SIP_RECVADDR variable.
1540 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1541 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1542 the ANI2/OLI information is set on the channel, which can be retrieved using
1543 the CALLERID function.
1545 * Peers can now be configured to support negotiation of ICE candidates using
1546 the setting icesupport. See res_rtp_asterisk changes for more information.
1548 * Added support for format attribute negotiation. See the Codecs changes for
1551 * Extra headers specified with SIPAddHeader are sent with the REFER message
1552 when using Transfer application. See refer_addheaders in sip.conf.sample.
1554 * Added support to use private party ID information with calls.
1556 * Adds an option discard_remote_hold_retrieval that when set stops telling
1557 the peer to start music on hold.
1562 * Added skinny version 17 protocol support.
1566 --------------------
1567 * Added ability to use multiple lines for a single phone. This allows multiple
1568 calls to occur on a single phone, using callwaiting and switching between calls.
1570 * Added option 'sharpdial' allowing end dialing by pressing # key
1572 * Added option 'interdigit_timer' to control phone dial timeout
1574 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1576 * Added global 'debug' option, that enables debug in channel driver
1578 * Added ability to translate on-screen menu in multiple languages. Tested on
1579 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1580 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1583 * In addition to English added French and Russian languages for on-screen menus
1585 * Reworked dialing number input: added dialing by timeout, immediate dial on
1586 on dialplan compare, phone number length now not limited by screen size
1588 * Added ability to pickup a call using features.conf defined value and
1594 * Add options namedcallgroup and namedpickupgroup to support installations
1595 where a higher number of groups (>64) is required.
1597 * Added support to use private party ID information with calls.
1602 * The minimum DTMF duration can now be configured in asterisk.conf
1603 as "mindtmfduration". The default value is (as before) set to 80 ms.
1604 (previously it was only available in source code)
1606 * Named ACLs can now be specified in acl.conf and used in configurations that
1607 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1608 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1609 working ACL. In addition, some CLI commands have been added to provide
1610 show information and allow for module reloading - see CLI Changes.
1612 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1613 items (separated by commas), and items in the rule can be negated by prefixing
1614 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1615 longer necessray to control the order that the 'permit' and 'deny' columns are
1616 returned from queries.
1618 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1619 be used within the dynamic weight attribute when specifying a mapping.
1621 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1622 header, instead of putting the user defined event name there. When enabled
1623 the UserDefType header is added for user defined events. This feature is
1624 enabled with the setting show_user_defined.
1626 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1627 line purposes use the following variables instead of their macro equivalents:
1628 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1629 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1630 cc_callback_macro in channel configurations.
1632 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1635 * Call files now support the "early_media" option to connect with an outgoing
1636 extension when early media is received.
1638 * Added support to use private party ID information with calls.
1643 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1644 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1645 AGI application would exit immediately after a channel hangup is detected.
1647 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1648 are resolved and each address is attempted in turn until one succeeds or
1652 AMI (Asterisk Manager Interface)
1654 * The originate action now has an option "EarlyMedia" that enables the
1655 call to bridge when we get early media in the call. Previously,
1656 early media was disregarded always when originating calls using AMI.
1658 * Added setvar= option to manager accounts (much like sip.conf)
1660 * Originate now generates an error response if the extension given is not found
1663 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1664 them if the i(variable) option is used. StopMixMonitor will accept
1665 MixMonitorID as an option to close specific MixMonitors.
1667 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1668 updated to include information about peers configured with
1669 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1670 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1671 returned if auto_force_rport is not enabled.
1673 * Added SIPpeerstatus manager command which will generate PeerStatus events
1674 similar to the existing PeerStatus events found in chan_sip on demand.
1676 * Hangup now can take a regular expression as the Channel option. If you want
1677 to hangup multiple channels, use /regex/ as the Channel option. Existing
1678 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1679 the manager will send you a list of channels back that were hung up.
1681 * Support for IPv6 addresses has been added.
1683 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1684 documentation is only generated when Asterisk is compiled using 'make full'.
1685 See the CLI section for commands to display AMI event information.
1687 * The AMI Hangup event now includes the AccountCode header so you can easily
1688 correlate with AMI Newchannel events.
1690 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1691 the StateInterface of the queue member.
1693 * Added AMI event SessionTimeout in the Call category that is issued when a
1694 call is terminated due to either RTP stream inactivity or SIP session timer
1697 * CEL events can now contain a user defined header UserDefType. See core
1698 changes for more information.
1700 * OOH323 ChannelUpdate events now contain a CallRef header.
1702 * Added PresenceState command. This command will report the presence state for
1703 the given presence provider.
1705 * Added Parkinglots command. This will list all parking lots as a series of
1706 AMI Parkinglot events.
1708 * Added MessageSend command. This behaves in the same manner as the
1709 MessageSend application, and is a technolgoy agnostic mechanism to send out
1710 of call text messages.
1712 * Added "message" class authorization. This grants an account permission to
1713 send out of call messages. Write-only.
1718 * The "dialplan add include" command has been modified to create context a context
1719 if one does not already exist. For instance, "dialplan add include foo into bar"
1720 will create context "bar" if it does not already exist.
1722 * A "dialplan remove context" command has been added to remove a context from
1725 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1726 filenames of all running mixmonitors on a channel.
1728 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1729 numeric instead of 0, 1, or 2.
1731 * "stun show status" will show a table describing how the STUN client is
1734 * "acl show [named acl]" will show information regarding a Named ACL. The
1735 acl module can be reloaded with "reload acl".
1737 * Added CLI command to display AMI event information - "manager show events",
1738 which shows a list of all known and documented AMI events, and "manager show
1739 event [event name]", which shows detail information about a specific AMI
1742 * The result of the CLI command "queue show" now includes the state interface
1743 information of the queue member.
1745 * The command "core set verbose" will now set a separate level of logging for
1746 each remote console without affecting any other console.
1748 * Added command "cdr show pgsql status" to check connection status
1750 * "sip show channel" will now display the complete route set.
1752 * Added "presencestate list" command. This command will list all custom
1753 presence states that have been set by using the PRESENCE_STATE dialplan
1756 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1757 command. This changes a custom presence to a new state.
1762 * Codec lists may now be modified by the '!' character, to allow succinct
1763 specification of a list of codecs allowed and disallowed, without the
1764 requirement to use two different keywords. For example, to specify all
1765 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1767 * Add support for parsing SDP attributes, generating SDP attributes, and
1768 passing it through. This support includes codecs such as H.263, H.264, SILK,
1769 and CELT. You are able to set up a call and have attribute information pass.
1770 This should help considerably with video calls.
1772 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1773 just like the GSM codec.
1777 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1778 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1782 * Asterisk version and build information is now logged at the beginning of a
1785 * Threads belonging to a particular call are now linked with callids which get
1786 added to any log messages produced by those threads. Log messages can now be
1787 easily identified as involved with a certain call by looking at their call id.
1788 Call ids may also be attached to log messages for just about any case where
1789 it can be determined to be related to a particular call.
1791 * Each logging destination and console now have an independent notion of the
1792 current verbosity level. Logger.conf now allows an optional argument to
1793 the 'verbose' specifier, indicating the level of verbosity sent to that
1794 particular logging destination. Additionally, remote consoles now each
1795 have their own verbosity level. The command 'core set verbose' will now set
1796 a separate level for each remote console without affecting any other
1802 * Added 'announcement' option which will play at the start of MOH and between
1803 songs in modes of MOH that can detect transitions between songs (eg.
1809 * New per parking lot options: comebackcontext and comebackdialtime. See
1810 configs/features.conf.sample for more details.
1812 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1815 * Channel variable PARKEDCALL is now set with the name of the parking lot
1816 when a timeout occurs.
1822 CDR Postgresql Driver
1824 * Added command "cdr show pgsql status" to check connection status
1827 CDR Adaptive ODBC Driver
1829 * Added schema option for databases that support specifying a schema.
1837 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1838 CALENDAR_WRITE has completed successfully.
1843 * A new option, 'probation' has been added to rtp.conf
1844 RTP in strictrtp mode can now require more than 1 packet to exit learning
1845 mode with a new source (and by default requires 4). The probation option
1846 allows the user to change the required number of packets in sequence to any
1847 desired value. Use a value of 1 to essentially restore the old behavior.
1848 Also, with strictrtp on, Asterisk will now drop all packets until learning
1849 mode has successfully exited. These changes are based on how pjmedia handles
1850 media sources and source changes.
1852 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1853 enabled or disabled using the icesupport setting. A variety of other
1854 settings have been introduced to configure STUN/TURN connections.
1859 * A new module, res_corosync, has been introduced. This module uses the
1860 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1861 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1862 Device State (presence) information. This module is very similar to, and
1863 is a replacement for the res_ais module that was in previous releases of
1869 * This module adds a cleaned up, drop-in replacement for res_jabber called
1870 res_xmpp. This provides the same externally facing functionality but is
1871 implemented differently internally. res_jabber has been deprecated in favor
1872 of res_xmpp; please see the UPGRADE.txt file for more information.
1877 * The safe_asterisk script has been updated to allow several of its parameters
1878 to be set from environment variables. This also enables a custom run
1879 directory of Asterisk to be specified, instead of defaulting to /tmp.
1881 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1882 its value to determine the directory to assume is the top-level directory of
1883 the source tree. If the variable is not set, it defaults to the current
1884 behavior and uses the current working directory.
1886 ------------------------------------------------------------------------------
1887 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1888 ------------------------------------------------------------------------------
1892 * Asterisk now has protocol independent support for processing text messages
1893 outside of a call. Messages are routed through the Asterisk dialplan.
1894 SIP MESSAGE and XMPP are currently supported. There are options in
1895 jabber.conf and sip.conf to allow enabling these features.
1896 -> jabber.conf: see the "sendtodialplan" and "context" options.
1897 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1898 and "outofcall_message_context" options.
1899 The MESSAGE() dialplan function and MessageSend() application have been
1900 added to go along with this functionality. More detailed usage information
1901 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1902 * If real-time text support (T.140) is negotiated, it will be preferred for
1903 sending text via the SendText application. For example, via SIP, messages
1904 that were once sent via the SIP MESSAGE request would be sent via RTP if
1905 T.140 text is negotiated for a call.
1909 * parkedmusicclass can now be set for non-default parking lots.
1911 Asterisk Manager Interface
1912 --------------------------
1913 * PeerStatus now includes Address and Port.
1914 * Added Hold events for when the remote party puts the call on and off hold
1915 for chan_dahdi ISDN channels.
1916 * Added new action MeetmeListRooms to list active conferences (shows same
1917 data as "meetme list" at the CLI).
1918 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1919 Description field that is set by 'description' in the channel configuration
1921 * Added Uniqueid header to UserEvent.
1922 * Added new action FilterAdd to control event filters for the current session.
1923 This requires the system permission and uses the same filter syntax as
1924 filters that can be defined in manager.conf
1925 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1926 versions had some instances of the event converted, but others were left
1927 as-is. All Unlink events should now be converted to Bridge events. The AMI
1928 protocol version number was incremented to 1.2 as a result of this change.
1930 Asterisk HTTP Server
1931 --------------------------
1932 * The HTTP Server can bind to IPv6 addresses.
1935 --------------------------
1936 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1937 with busydetect. usage example: busypattern=200,200,200,600
1940 --------------------------
1941 * New 'gtalk show settings' command showing the current settings loaded from
1943 * The 'logger reload' command now supports an optional argument, specifying an
1944 alternate configuration file to use.
1945 * 'dialplan add extension' command will now automatically create a context if
1946 the specified context does not exist with a message indicated it did so.
1947 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1948 Description field which can be populated with 'description' in the channel
1949 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1952 --------------------------
1953 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1954 thus allowing records which do NOT match the specified filter.
1955 * Added ability to log CONGESTION calls to CDR
1958 --------------------------
1959 * Ability to define custom SILK formats in codecs.conf.
1960 * Addition of speex32 audio format with translation.
1961 * CELT codec pass-through support and ability to define
1962 custom CELT formats in codecs.conf.
1963 * Ability to read raw signed linear files with sample rates
1964 ranging from 8khz - 192khz. The new file extensions introduced
1965 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1966 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1967 Skinny, H.323, etc) can still only support the following codecs:
1968 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1969 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1970 Video: h261, h263, h263p, h264, mpeg4
1975 --------------------------
1976 * New highly optimized and customizable ConfBridge application capable of
1977 mixing audio at sample rates ranging from 8khz-96khz.
1978 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1979 and bridge profiles on a channel.
1980 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1981 about a conference such as locked status and number of parties, admins,
1983 * Addition of video_mode option in confbridge.conf for adding video support
1984 into a bridge profile.
1985 * Addition of the follow_talker video_mode in confbridge.conf. This video
1986 mode dynamically switches the video feed to always display the loudest talker
1987 supplying video in the conference.
1991 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1992 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1993 variables from asterisk.conf.
1997 * Addition of the JITTERBUFFER dialplan function. This function allows
1998 for jitterbuffering to occur on the read side of a channel. By using
1999 this function conference applications such as ConfBridge and MeetMe can
2000 have the rx streams jitterbuffered before conference mixing occurs.
2001 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2003 * Added STRREPLACE function. This function let's the user search a variable
2004 for a given string to replace with another string as many times as the
2005 user specifies or just throughout the whole string.
2006 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2007 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2008 * Added extensions to chan_ooh323 in function CHANNEL()
2010 libpri channel driver (chan_dahdi) DAHDI changes
2011 --------------------------
2012 * Added moh_signaling option to specify what to do when the channel's bridged
2013 peer puts the ISDN channel on hold.
2014 * Added display_send and display_receive options to control how the display ie
2015 is handled. To send display text from the dialplan use the SendText()
2016 application when the option is enabled.
2017 * Added mcid_send option to allow sending a MCID request on a span.
2020 --------------------------
2021 * Added setvar option to calendar.conf to allow setting channel variables on
2022 notification channels.
2023 * Added "calendar show types" CLI command to list registered calendar
2027 --------------------------
2028 * Added two new options, r and t with file name arguments to record
2029 single direction (unmixed) audio recording separate from the bidirectional
2030 (mixed) recording. The mixed file name argument is optional now as long
2031 as at least one recording option is used.
2034 --------------------------
2035 * Added a new option, l, which will disable local call optimization for
2036 channels involved with the FollowMe thread. Use this option to improve
2037 compatability for a FollowMe call with certain dialplan apps, options, and
2041 --------------------------
2042 * Added option "k" that will automatically close the conference when there's
2043 only one person left when a user exits the conference.
2046 --------------------------
2047 * cel_pgsql now supports the 'extra' column for data added using the
2048 CELGenUserEvent() application.
2051 --------------------------
2052 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2053 in the sample extensions.lua file for syntax details.
2054 * Applications that perform jumps in the dialplan such as Goto will now
2055 execute properly. When pbx_lua detects that the context, extension, or
2056 priority we are executing on has changed it will immediately return control
2057 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2058 the priority after the currently executing priority.
2059 * An autoservice is now started by default for pbx_lua channels. It can be
2060 stopped and restarted using the autoservice_stop() and autoservice_start()
2064 --------------------------
2065 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2066 into a FAXStatus event with an 'Operation' header that will be either
2067 'send', 'receive', and 'gateway'.
2068 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2069 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2070 feature will handle converting a fax call between an audio T.30 fax terminal
2071 and an IFP T.38 fax terminal.
2075 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2076 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2077 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2081 * Added general option negative_penalty_invalid default off. when set
2082 members are seen as invalid/logged out when there penalty is negative.
2083 for realtime members when set remove from queue will set penalty to -1.
2084 * Added queue option autopausedelay when autopause is enabled it will be
2085 delayed for this number of seconds since last successful call if there
2086 was no prior call the agent will be autopaused immediately.
2087 * Added member option ignorebusy this when set and ringinuse is not
2088 will allow per member control of multiple calls as ringinuse does for
2093 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2095 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2096 one participant left (much like a normal call bridge)
2097 * Added extra argument to Originate to set timeout.
2101 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2102 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2103 utility in the UTILS section of menuselect. If an existing astdb is found and no
2104 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2105 convert an existing astdb to the SQLite3 version automatically at runtime.
2109 * Modules marked as deprecated are no longer marked as building by default. Enabling
2110 these modules is still available via menuselect.
2114 * authdebug is now disabled by default. To enable this functionaility again
2115 set authdebug = yes in iax.conf.
2119 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2120 releases it was disabled.
2124 * The PBX core previously made a call with a non-existing extension test for
2125 extension s@default and jump there if the extension existed.
2126 This was a bad default behaviour and violated the principle of least surprise.
2127 It has therefore been changed in this release. It may affect some
2128 applications and configurations that rely on this behaviour. Most channel
2129 drivers have avoided this for many releases by testing whether the extension
2130 called exists before starting the PBX and generating a local error.
2131 This behaviour still exists and works as before.
2133 Extension "s" is used when no extension is given in a channel driver,
2134 like immediate answer in DAHDI or calling to a domain with no user part
2137 ------------------------------------------------------------------------------
2138 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2139 ------------------------------------------------------------------------------
2143 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2144 now defaults to force_rport. It is very important that phones requiring nat=no be
2145 specifically set as such instead of relying on the default setting. If at all
2146 possible, all devices should have nat settings configured in the general section as
2147 opposed to configuring nat per-device.
2148 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2149 codecs sent in response to an INVITE to the single most preferred codec.
2150 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2151 to be used for the outgoing call. It must be one of the codecs configured
2153 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2154 to be used for holding a private key. If tlsprivatekey is not specified,
2155 tlscertfile is searched for both public and private key.
2156 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2157 outbound client connections to be specified.
2158 * The sendrpid parameter has been expanded to include the options
2159 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2160 header to be sent (equivalent to setting sendrpid=yes) and setting
2161 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2162 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2163 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2164 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2165 will accept the SDP even if the SDP version number is not properly incremented,
2166 but will generate a warning in the log indicating that the SIP peer that sent
2167 the SDP should have the 'ignoresdpversion' option set.
2168 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2169 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2170 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2171 remote side requests it and disables symmetric RTP support. Setting it to
2172 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2173 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2174 and enables symmetric RTP support.
2175 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2176 response. This permits the master channel to know how each channel dialled
2177 in a multi-channel setup resolved in an individual way. This carries a
2178 performance penalty and can be disabled in sip.conf using the
2179 'storesipcause' option.
2180 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2181 configuration for the externip and externhost options when tcp or tls is used.
2182 * Added support for message body (stored in content variable) to SIP NOTIFY message
2183 accessible via AMI and CLI.
2184 * Added 'media_address' configuration option which can be used to explicitly specify
2185 the IP address to use in the SDP for media (audio, video, and text) streams.
2186 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2187 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2189 * Added 'use_q850_reason' configuration option for generating and parsing
2190 if available Reason: Q.850;cause=<cause code> header. It is implemented
2191 in some gateways for better passing PRI/SS7 cause codes via SIP.
2192 * When dialing SIP peers, a new component may be added to the end of the dialstring
2193 to indicate that a specific remote IP address or host should be used when dialing
2194 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2195 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2196 ability to selectively force bridged channels to also be encrypted is also
2197 implemented. Branching in the dialplan can be done based on whether or not
2198 a channel has secure media and/or signaling.
2199 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2201 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2202 Charge messages to snom phones.
2203 * Added support for G.719 media streams.
2204 * Added support for 16khz signed linear media streams.
2205 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2206 RTP has been outfitted with the same abilities.
2207 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2208 available in device configurations as well as in the dial plan.
2209 * Addition of the 'subscribe_network_change' option for turning on and off
2210 res_stun_monitor module support in chan_sip.
2211 * Addition of the 'auth_options_requests' option for turning on and off
2212 authentication for OPTIONS requests in chan_sip.
2216 * Add #tryinclude statement for config files. This provides the same
2217 functionality as the #include statement however an asterisk module will
2218 still load if the filename does not exist. Using the #include statement
2219 Asterisk will not allow the module to load.
2223 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2224 on realtime updates.
2225 * Added the ability for chan_iax2 to inform the dialplan whether or not
2226 encryption is being used. This interoperates with the SIP SRTP implementation
2227 so that a secure SIP call can be bridged to a secure IAX call when the
2228 dialplan requires bridged channels to be "secure".
2229 * Addition of the 'subscribe_network_change' option for turning on and off
2230 res_stun_monitor module support in chan_iax.
2235 * Added ability to preset channel variables on indicated lines with the setvar
2236 configuration option. Also, clearvars=all resets the list of variables back
2238 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2239 See configs/res_pktccops.conf for more information.
2241 XMPP Google Talk/Jingle changes
2242 -------------------------------
2243 * Added the externip option to gtalk.conf.
2244 * Added the stunaddr option to gtalk.conf which allows for the automatic
2245 retrieval of the external ip from a stun server.
2249 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2250 match to a partial channel name.
2251 * Added .m3u support for Mp3Player application.
2252 * Added progress option to the app_dial D() option. When progress DTMF is
2253 present, those values are sent immediately upon receiving a PROGRESS message
2254 regardless if the call has been answered or not.
2255 * Added functionality to the app_dial F() option to continue with execution
2256 at the current location when no parameters are provided.
2257 * Added the 'a' option to app_dial to answer the calling channel before any
2258 announcements or macros are executed.
2259 * Modified app_dial to set answertime when the called channel answers even if
2260 the called channel hangs up during playback of an announcement.
2261 * Modified app_dial 'r' option to support an additional parameter to play an
2262 indication tone from indications.conf
2263 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2264 to cycle through the next available channel. By default this is still '*'.
2265 * Added x() option to app_chanspy. This option allows DTMF to be set to
2266 exit the application.
2267 * The Voicemail application has been improved to automatically ignore messages
2268 that only contain silence.
2269 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2270 associated mailbox(es) to be greetings-only.
2271 * The ChanSpy application now has the 'S' option, which makes the application
2272 automatically exit once it hits a point where no more channels are available
2274 * The ChanSpy application also now has the 'E' option, which spies on a single
2275 channel and exits when that channel hangs up.
2276 * The MeetMe application now turns on the DENOISE() function by default, for
2277 each participant. In our tests, this has significantly decreased background
2278 noise (especially noisy data centers).
2279 * Voicemail now permits storage of secrets in a separate file, located in the
2280 spool directory of each individual user. The control for this is located in
2281 the "passwordlocation" option in voicemail.conf. Please see the sample
2282 configuration for more information.
2283 * The ChanIsAvail application now exposes the returned cause code using a separate
2284 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2285 * Added 'd' option to app_followme. This option disables the "Please hold"
2287 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2288 received will terminate recording.
2289 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2290 Previously the folder could only be set per context, but has now been extended
2291 using the imapfolder option.
2292 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2293 * Voicemail now allows the pager date format to be specified separately from the
2295 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2296 to allow joining, leaving, and sending text to group chats.
2297 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2298 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2299 to all paged phones (and optionally excluding the caller's one using the new
2300 option 'n') before the call is bridged.
2301 * The 'f' option to Dial has been augmented to take an optional argument. If no
2302 argument is provided, the 'f' option works as it always has. If an argument is
2303 provided, then the connected party information of all outgoing channels created
2304 during the Dial will be set to the argument passed to the 'f' option.
2305 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2307 * The OSP lookup application adds in/outbound network ID, optional security,
2308 number portability, QoS reporting, destination IP port, custom info and service
2310 * Added new application VMSayName that will play the recorded name of the voicemail
2311 user if it exists, otherwise will play the mailbox number.
2312 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2313 retrieve state for a particular bridge, where <name> is the conference name
2314 * app_directory now allows exiting at any time using the operator or pound key.
2315 * Voicemail now supports setting a locale per-mailbox.
2316 * Two new applications are provided for declining counting phrases in multiple
2317 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2319 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2321 * Voicemail now includes rdnis within msgXXXX.txt file.
2322 * ExternalIVR now supports IPv6 addresses.
2323 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2324 at https://wiki.asterisk.org/wiki/x/oQBB
2325 * ParkedCall and Park can now specify the parking lot to use.
2329 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2330 over SRV records associated with a specific service. From the CLI, type
2331 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2332 details on how these may be used.
2333 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2334 pitch of a channel's tx and rx audio streams.
2335 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2336 setting various connected line and redirecting party information.
2337 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2338 support ISDN subaddressing.
2339 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2340 * For DAHDI channels, the CHANNEL() dialplan function now allows
2341 the dialplan to request changes in the configuration of the active
2342 echo canceller on the channel (if any), for the current call only.
2345 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2347 The possible values are:
2349 on - normal mode (the echo canceller is actually reinitialized)
2351 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2353 voice - voice mode (returns from FAX mode, reverting the changes that
2354 were made when FAX mode was requested)
2355 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2356 and setting variables on the channel which created the current channel.
2357 Administrators should take care to avoid naming conflicts, when multiple
2358 channels are dialled at once, especially when used with the Local channel
2359 construct (which all could set variables on the master channel). Usage
2360 of the HASH() dialplan function, with the key set to the name of the slave
2361 channel, is one approach that will avoid conflicts.
2362 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2364 * func_odbc now allows multiple row results to be retrieved without using
2365 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2366 from the same query by using the name of the function which retrieved the
2367 first row as an argument to ODBC_FETCH().
2368 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2369 dialplan. This function returns the content of the received message.
2370 * Added REPLACE, which searches a given variable name for a set of characters,
2371 then either replaces them with a single character or deletes them.
2372 * Added PASSTHRU, which literally passes the same argument back as its return
2373 value. The intent is to be able to use a literal string argument to
2374 functions that currently require a variable name as an argument.
2375 * HASH-associated variables now can be inherited across channel creation, by
2376 prefixing the name of the hash at assignment with the appropriate number of
2377 underscores, just like variables.
2378 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2379 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2380 whether or not channels that are bridged to the current channel will be
2381 required to have secure signaling and/or media.
2382 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2383 the current channel has secure signaling and/or media.
2384 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2385 "no_media_path" option.
2386 Returns "0" if there is a B channel associated with the call.
2387 Returns "1" if no B channel is associated with the call. The call is either
2388 on hold or is a call waiting call.
2389 * Added option to dialplan function CDR(), the 'f' option
2390 allows for high resolution times for billsec and duration fields.
2391 * FILE() now supports line-mode and writing.
2392 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2393 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2397 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2398 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2399 and is set when a dynamic feature is triggered.
2400 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2401 to dynamically create a new parking lot matching the value this varible is
2403 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2404 features.conf that should be the base for dynamic parkinglots.
2405 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2406 parkinglot should have.
2407 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2408 parkinglot should have.
2409 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2414 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2415 timeout has expired.
2416 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2417 to the caller when an Agent's phone is ringing. This can be used to indicate
2418 to the caller that their call is about to be picked up, which is nice when
2419 one has been on hold for an extened period of time.
2420 * A new config option, penaltymemberslimit, has been added to queues.conf.
2421 When set this option will disregard penalty settings when a queue has too
2423 * A new option, 'I' has been added to both app_queue and app_dial.
2424 By setting this option, Asterisk will not update the caller with
2425 connected line changes or redirecting party changes when they occur.
2426 * A 'relative-periodic-announce' option has been added to queues.conf. When
2427 enabled, this option will cause periodic announce times to be calculated
2428 from the end of announcements rather than from the beginning.
2429 * The autopause option in queues.conf can be passed a new value, "all." The
2430 result is that if a member becomes auto-paused, he will be paused in all
2431 queues for which he is a member, not just the queue that failed to reach
2433 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2434 * The queue logger now allows events to optionally propagate to a file,
2435 even when realtime logging is turned on. Additionally, realtime logging
2436 supports sending the event arguments to 5 individual fields, although it
2437 will fallback to the previous data definition, if the new table layout is
2440 mISDN channel driver (chan_misdn) changes
2441 ----------------------------------------
2442 * Added display_connected parameter to misdn.conf to put a display string
2443 in the CONNECT message containing the connected name and/or number if
2444 the presentation setting permits it.
2445 * Added display_setup parameter to misdn.conf to put a display string
2446 in the SETUP message containing the caller name and/or number if the
2447 presentation setting permits it.
2448 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2449 indicate the dialplan settings are to be obtained from the asterisk
2451 * Made misdn.conf parameter callerid accept the "name" <number> format
2452 used by the rest of the system.
2453 * Made use the nationalprefix and internationalprefix misdn.conf
2454 parameters to prefix any received number from the ISDN link if that
2455 number has the corresponding Type-Of-Number. NOTE: This includes
2456 comparing the incoming call's dialed number against the MSN list.
2457 * Added the following new parameters: unknownprefix, netspecificprefix,
2458 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2459 received number from the ISDN link if that number has the corresponding
2461 * Added new dialplan application misdn_command which permits controlling
2462 the CCBS/CCNR functionality.
2463 * Added new dialplan function mISDN_CC which permits retrieval of various
2464 values from an active call completion record.
2465 * For PTP, you should manually send the COLR of the redirected-to party
2466 for an incomming redirected call if the incoming call could experience
2467 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2468 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2469 if the REDIRECTING(from-num) is not empty.
2470 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2471 option on all of the REDIRECTING statements before dialing the
2472 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2473 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2474 redirecting-to presentation (COLR) when it becomes available.
2475 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2478 thirdparty mISDN enhancements
2479 -----------------------------
2480 mISDN has been modified by Digium, Inc. to greatly expand facility message
2482 * Enhanced COLP support for call diversion and transfer.
2483 * CCBS/CCNR support.
2485 The latest modified mISDN v1.1.x based version is available at:
2486 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2487 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2489 Tagged versions of the modified mISDN code are available under:
2490 http://svn.digium.com/svn/thirdparty/mISDN/tags
2491 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2493 libpri channel driver (chan_dahdi) DAHDI changes
2494 -------------------------------------------
2495 * The channel variable PRIREDIRECTREASON is now just a status variable
2496 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2497 to read and alter the reason.
2498 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2499 redirected-to party for an incomming redirected call if the incoming call
2500 could experience further redirects. Just set the
2501 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2502 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2504 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2505 use the inhibit(i) option on all of the REDIRECTING statements before
2506 dialing the redirected-to party. You still have to set the
2507 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2508 will update the redirecting-to presentation (COLR) when it becomes available.
2509 * Added the ability to ignore calls that are not in a Multiple Subscriber
2510 Number (MSN) list for PTMP CPE interfaces.
2511 * Added dynamic range compression support for dahdi channels. It is
2512 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2513 * Added support for ISDN calling and called subaddress with partial support
2514 for connected line subaddress.
2515 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2516 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2517 to transfer a held call on disconnect similar to an analog phone.
2518 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2519 Will reroute/deflect an outgoing call when receive the message.
2520 Can use the DAHDISendCallreroutingFacility to send the message for the
2522 * Added standard location to add options to chan_dahdi dialing:
2523 Dial(DAHDI/g1[/extension[/options]])
2526 R Reverse charging indication
2527 * Added Reverse Charging Indication (Collect calls) send/receive option.
2528 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2529 Dial(DAHDI/g1/extension/R)
2530 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2531 (requires latest LibPRI)
2532 * Added ability to send/receive keypad digits in the SETUP message.
2533 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2534 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2535 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2536 (requires latest LibPRI)
2537 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2538 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2539 back into the same interface. Tromboned calls happen because of call routing,
2540 call deflection, call forwarding, and call transfer.
2541 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2542 * Added the ability to support call waiting calls. (The SETUP has no B channel
2544 * Added Malicious Call ID (MCID) event to the AMI call event class.
2545 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2547 Asterisk Manager Interface
2548 --------------------------
2549 * The Hangup action now accepts a Cause header which may be used to
2550 set the channel's hangup cause.
2551 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2552 to specify a separate .pem file to hold a private key. By default sslcert
2553 is used to hold both the public and private key.
2554 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2555 for options containing the 'tls' prefix. For example, 'sslenable' is now
2556 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2557 across all .conf files. All affected sample.conf files have been modified to
2558 reflect this change. Previous options such as 'sslenable' still work,
2559 but options with the 'tls' prefix are preferred.
2560 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2561 in a channel. (res_mutestream.so)
2562 * The configuration file manager.conf now supports a channelvars option, which
2563 specifies a list of channel variables to include in each channel-oriented
2565 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2566 and ExtraPriority to allow redirecting the second channel to a different
2567 location than the first.
2568 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2570 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2571 in a MixMonitor recording.
2572 * The 'iax2 show peers' output is now similar to the expected output of
2574 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2576 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2577 AOC-E messages on a channel.
2578 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2579 conform more closely to similar events.
2580 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2582 * Added optional parkinglot variable for park command.
2583 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2584 if CallerIDNum and CallerIDName headers are also present.
2586 Channel Event Logging
2587 ---------------------
2588 * A new interface, CEL, is introduced here. CEL logs single events, much like
2589 the AMI, but it differs from the AMI in that it logs to db backends much
2590 like CDR does; is based on the event subsystem introduced by Russell, and
2591 can share in all its benefits; allows multiple backends to operate like CDR;
2592 is specialized to event data that would be of concern to billing sytems,
2593 like CDR. Backends for logging and accounting calls have been produced,
2594 but a new CDR backend is still in development.
2598 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2599 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2600 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2601 * Multiple files and formats can now be specified in cdr_custom.conf.
2602 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2603 See configs/cdr_syslog.conf.sample for more information.
2604 * A 'sequence' field has been added to CDRs which can be combined with
2605 linkedid or uniqueid to uniquely identify a CDR.
2606 * Handling of billsec and duration field has changed. If your table definition
2607 specifies those fields as float,double or similar they will now be logged with
2608 microsecond accuracy instead of a whole integer.
2610 Calendaring for Asterisk
2611 ------------------------
2612 * A new set of modules were added supporing calendar integration with Asterisk.
2613 Dialplan functions for reading from and writing to calendars are included,
2614 as well as the ability to execute dialplan logic upon calendar event notifications.
2615 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2616 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2617 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2618 2003 support does not support forms-based authentication).
2620 Call Completion Supplementary Services for Asterisk
2621 ---------------------------------------------------
2622 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2623 DAHDI/ISDN supports call completion for the following switch types:
2624 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2625 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2627 Multicast RTP Support
2628 ---------------------
2629 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2630 The channel driver can be used with the Page application to perform multicast RTP
2631 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2632 Type can be either basic or linksys.
2633 Destination is the IP address and port for the RTP packets.
2634 Control address is specific to the linksys type and is used for sending the control
2635 packets unique to them.
2637 Security Events Framework
2638 -------------------------
2639 * Asterisk has a new C API for reporting security events. The module res_security_log
2640 sends these events to the "security" logger level. Currently, AMI is the only
2641 Asterisk component that reports security events. However, SIP support will be
2642 coming soon. For more information on the security events framework, see the
2643 "Asterisk Security Framework" section of the Asterisk wiki at
2644 https://wiki.asterisk.org/wiki/x/wgBQ
2645 * SIP support was added in Asterisk 10
2646 * This API now supports IPv6 addresses
2650 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2651 * A spandsp based fax backend (res_fax_spandsp) has been added.
2652 * The app_fax module has been deprecated in favor of the res_fax module and
2653 the new res_fax_spandsp backend.
2654 * The SendFAX and ReceiveFAX applications now send their log messages to a
2655 'fax' logger level, instead of to the generic logger levels. To see these
2656 messages, the system's logger.conf file will need to direct the 'fax' logger
2657 level to one or more destinations; the logger.conf.sample file includes an
2658 example of how to do this. Note that if the 'fax' logger level is *not*
2659 directed to at least one destination, log messages generated by these
2660 applications will be lost, and that if the 'fax' logger level is directed to
2661 the console, the 'core set verbose' and 'core set debug' CLI commands will
2662 have no effect on whether the messages appear on the console or not.
2666 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2667 Now, in order to enable transmitting silence during record the transmit_silence
2668 option should be used. transmit_silence_during_record remains a valid option, but
2669 defaults to the behavior of the transmit_silence option.
2670 * Addition of the Unit Test Framework API for managing registration and execution
2671 of unit tests with the purpose of verifying the operation of C functions.
2672 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2673 XMPP text messages to the remote JID.
2674 * Modules.conf has a new option - "require" - that marks a module as critical for
2675 the execution of Asterisk.
2676 If one of the required modules fail to load, Asterisk will exit with a return
2678 * An 'X' option has been added to the asterisk application which enables #exec support.
2679 This allows #exec to be used in asterisk.conf.
2680 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2681 * A new lockconfdir option has been added to asterisk.conf to protect the
2682 configuration directory (/etc/asterisk by default) during reloads.
2683 * The parkeddynamic option has been added to features.conf to enable the creation
2684 of dynamic parkinglots.
2685 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2686 the reportalarms config option.
2687 * chan_dahdi supports dialing configuring and dialing by device file name.
2688 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2689 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2690 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2691 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2692 Handy for the above name-based syntax as it does not depend on
2693 initialization order.
2694 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2695 significant increase in performance (about 3X) for installations using this switchtype.
2696 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2697 AIS. For more information, please see the Distributed Device State section of the
2698 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2699 * The addition of G.719 pass-through support.
2700 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2701 during device configuration.
2702 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2703 have less than 3 lines on the LCD.
2704 * Realtime now supports database failover. See the sample extconfig.conf for details.
2705 * The addition of improved translation path building for wideband codecs. Sample
2706 rate changes during translation are now avoided unless absolutely necessary.
2707 * The addition of the res_stun_monitor module for monitoring and reacting to network
2708 changes while behind a NAT.
2709 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2710 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2711 These allow support for any Administration. Default is AT&T values.
2715 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2716 optionally accept a filename, to apply the setting only to the code generated from
2717 that source file when Asterisk was built. However, there are some modules in Asterisk
2718 that are composed of multiple source files, so this did not result in the behavior
2719 that users expected. In this version, 'core set debug' and 'core set verbose'
2720 can optionally accept *module* names instead (with or without the .so extension),
2721 which applies the setting to the entire module specified, regardless of which source
2722 files it was built from.
2723 * New 'manager show settings' command showing the current settings loaded from
2725 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2726 the channel hangup request to all channels.
2727 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2729 ------------------------------------------------------------------------------
2730 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2731 ------------------------------------------------------------------------------
2735 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2736 Snom phones use this for call pickup of extensions that the phone is
2738 * Added support for setting the domain in the URI for caller of an
2739 outbound call by using the SIPFROMDOMAIN channel variable.
2740 * Added a new configuration option "remotesecret" for authentication to
2741 remote services. For backwards compatibility, "secret" still has the
2742 same function as before, but now you can configure both a remote secret and a
2743 local secret for mutual authentication.
2744 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2745 the sound will be played to the target of an attended transfer
2746 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2747 finer control over how many peers Asterisk will qualify and the gap between them
2748 when all peers need to be qualified at the same time.
2749 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2750 (either globally or for a specific peer), chan_sip will treat any SDP data
2751 it receives as new data and update the media stream accordingly. By
2752 default, Asterisk will only modify the media stream if the SDP session
2753 version received is different from the current SDP session version. This
2754 option is required to interoperate with devices that have non-standard SDP
2755 session version implementations (observed with Microsoft OCS). This option
2756 is disabled by default.
2757 * The parsing of register => lines in sip.conf has been modified to allow a port
2758 to be present in the "user" portion. Please see the sip.conf.sample file for more
2760 * Added support for subscribing to MWI on a remote server and making the status available
2761 as a mailbox. Please see the sip.conf.sample file for more information.
2762 * Added a function to remove SIP headers added in the dialplan before the
2763 first INVITE is generated - SIPRemoveHeader()
2764 * Channel variables set with setvar= in a device configuration is now
2765 set both for inbound and outbound calls.
2766 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2770 * Added immediate option to iax.conf
2771 * Added forceencryption option to iax.conf
2772 * Added Encryption and Trunk status to manager command "iaxpeers"
2776 * The configuration file now holds separate sections for devices and lines.
2777 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2782 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2783 support for LibOpenR2. http://www.libopenr2.org/
2784 * The UK option waitfordialtone has been added for use with BT analog
2786 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2787 is used in conjunction with the 'faxdetect' configuration option. When
2788 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2789 switch to the configured faxbuffers policy. For example, to use 6 buffers
2790 and a 'full' buffer policy for a fax transmission, add:
2792 The faxbuffers configuration will be in affect until the call is torn down.
2793 * Added service message support for 4ESS/5ESS switches.
2797 * For DAHDI channels, the CHANNEL() dialplan function now
2798 supports changing the channel's buffer policy (for the current
2799 call only), using this syntax:
2801 exten => s,n,Set(CHANNEL(buffers)=6,full)
2803 This would change the channel to the 'full' buffer policy and
2804 6 (six) buffers. Possible options for this setting are the same
2805 as those in chan_dahdi.conf.
2806 * Added a new dialplan function, CURLOPT, which permits setting various
2807 options that may be useful with the CURL dialplan function, such as
2808 cookies, proxies, connection timeouts, passwords, etc.
2809 * Permit the syntax and synopsis fields of the corresponding dialplan
2810 functions to be individually set from func_odbc.conf.
2811 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2812 * func_odbc now may specify an insert query to execute, when the write query
2813 affects 0 rows (usually indicating that no such row exists).
2814 * Added a new dialplan function, LISTFILTER, which permits removing elements
2815 from a set list, by name. Uses the same general syntax as the existing CUT
2816 and FIELDQTY dialplan functions, which also manage lists.
2817 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2818 obtaining realtime data from the dialplan.
2819 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2820 a subroutine when using the GoSub() and Return() applications.
2821 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2822 of "core show function AUDIOHOOK_INHERIT" from the CLI
2823 * Added AES_ENCRYPT. For information on its use, please see the output
2824 of "core show function AES_ENCRYPT" from the CLI
2825 * Added AES_DECRYPT. For information on its use, please see the output
2826 of "core show function AES_DECRYPT" from the CLI
2827 * func_odbc now supports database transactions across multiple queries.
2831 * Scheduled meetme conferences may now have their end times extended by
2833 * app_authenticate now gives the ability to select a prompt other than
2835 * app_directory now pays attention to the searchcontexts setting in
2836 voicemail.conf and will look through all contexts, if no context is
2837 specified in the initial argument.
2838 * A new application, Originate, has been introduced, that allows asynchronous
2839 call origination from the dialplan.
2840 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2841 in addition to the setting in the "general" context.
2842 * Added ConfBridge dialplan application which does conference bridges without
2843 DAHDI. For information on its use, please see the output of
2844 "core show application ConfBridge" from the CLI.
2848 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2849 operation to the AMI Redirect action.
2850 * extensions.conf now allows you to use keyword "same" to define an extension
2851 without actually specifying an extension. It uses exactly the same pattern
2852 as previously used on the last "exten" line. For example:
2853 exten => 123,1,NoOp(something)
2854 same => n,SomethingElse()
2855 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2856 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2857 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2858 by the new clialiases module. See cli_aliases.conf.sample file.
2859 * Times within timespecs are now accurate down to the minute. This is a change
2860 from historical Asterisk, which only provided timespecs rounded to the nearest
2861 even (read: evenly divisible by 2) minute mark.
2862 * The realtime switch now supports an option flag, 'p', which disables searches for
2864 * In addition to a time range and date range, timespecs now accept a 5th optional
2865 argument, timezone. This allows you to perform time checks on alternate
2866 timezones, especially if those daylight savings time ranges vary from your
2867 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2869 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2870 give you the correct output for an asterisk box behind nat. It will give you the
2871 externhost and localnet settings.
2872 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2873 can connect calls in passthrough mode, as well as record and play back files.
2874 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2875 using pickupsound and pickupfailsound in features.conf.
2876 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2877 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2878 instead of the /var/run/asterisk.pid where it used to be. This will make
2879 installs as non-root easier to manage.
2884 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2885 be written; they will no longer be explicitly written.
2887 Asterisk Manager Interface
2888 --------------------------
2889 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2890 a non-empty value) in your request. If you do this, any pending AMI events will
2891 *not* be included in the response to your request as they would normally, but
2892 will be left in the event queue for the next request you make to retrieve. For
2893 some applications, this will allow you to guarantee that you will only see
2894 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2895 To know whether the Asterisk server supports this header or not, your client can
2896 inspect the first response back from the server to see if it includes this header:
2898 Pragma: SuppressEvents
2900 If this is included, the server supports event suppression.
2902 * Added 4 new Actions to list skinny device(s) and line(s)
2908 LDAP Schema File Additions
2909 --------------------------
2910 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2911 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2913 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2914 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2915 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2916 * Removed redundant IPaddr (there's already IPAddress)
2917 - Gives more configuration Flags for SIP-Users available (tested)
2918 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2919 without extensibleObject (which really should be the last resort); gives
2920 also additional possibilities for LDAP-filter
2922 ------------------------------------------------------------------------------
2923 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2924 ------------------------------------------------------------------------------
2926 Device State Handling
2927 ---------------------
2928 * The event infrastructure in Asterisk got another big update to help support
2929 distributed events. It currently supports distributed device state and
2930 distributed Voicemail MWI (Message Waiting Indication). A new module has
2931 been merged, res_ais, which facilitates communicating events between servers.
2932 It uses the SAForum AIS (Service Availability Forum Application Interface
2933 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2934 a cluster of Asterisk servers, and to share events between them. For more
2935 information on setting this up, refer to the Distributed Device State section
2936 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2940 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2941 variables from an Asterisk configuration file.
2942 * The JACK_HOOK function now has a c() option to supply a custom client name.
2943 * Added two new dialplan functions from libspeex for audio gain control and
2944 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2945 rx directions of a channel from the dialplan.
2946 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2947 based on other parameters. The default is still to search based on the
2948 forwarding station ID. However, there are new options that allow you to search
2949 based on the message desk terminal ID, or the message desk number.
2950 * TIMEOUT() has been modified to be accurate down to the millisecond.
2951 * ENUM*() functions now include the following new options:
2952 - 'u' returns the full URI and does not strip off the URI-scheme.
2953 - 's' triggers ISN specific rewriting
2954 - 'i' looks for branches into an Infrastructure ENUM tree
2955 - 'd' for a direct DNS lookup without any flipping of digits.
2956 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2957 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2958 deviation of jitter, rtt, and loss for a call using chan_sip.
2960 DAHDI channel driver (chan_dahdi) Changes
2961 ----------------------------------------
2962 * Channels can now be configured using named sections in chan_dahdi.conf, just
2963 like other channel drivers, including the use of templates.
2964 * The default for pridialplan has changed from 'national' to 'unknown'.
2968 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2969 to something that matches the pattern a hint will be created using the contents
2970 and variables evaluated.
2971 * Dialplan matching has been extended to allow an extension to return to the
2972 PBX core to wait for more digits. This is done by using the new dialplan
2973 application called "Incomplete". This will permit a whole new level of
2974 extension control, by giving the administrator more control over early
2975 matches employing one of the short-circuit pattern match operators. Note
2976 that custom applications can trigger this same behavior by returning the
2977 special value AST_PBX_INCOMPLETE.
2981 * Directory now permits both first and last names to be matched at the same
2982 time. In addition, the number of digits to enter of the name can be set in
2983 the arguments to Directory; previously, you could enter only 3, regardless
2984 of how many names are in your company. For large companies, this should be
2986 * Voicemail now permits a mailbox setting to wrap around from first to last
2987 messages, if the "messagewrap" option is set to a true value.
2988 * Voicemail now permits an external script to be run, for password validation.
2989 The script should output "VALID" or "INVALID" on stdout, depending upon the
2990 wish to validate or invalidate the password given. Arguments are:
2991 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2993 * Dial has a new option: F(context^extension^pri), which permits a callee to
2994 continue in the dialplan, at the specified label, if the caller hangs up.
2995 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2996 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2997 * The Jack application now has a c() option to supply a custom client name.
2998 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2999 like the pre-existing whisper mode, except that the spy can also talk to the
3000 participant on the bridged channel as well.
3001 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3002 to be spoken instead of the channel name or number. For more information on the
3003 use of this option, issue the command "core show application ChanSpy" from the
3005 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3006 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3007 words, if using the 'd' option, it is not possible to enter a number to append to
3008 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3009 change to whisper mode, and pressing 6 will change to barge mode.
3010 * ExternalIVR now takes several options that affect the way it performs, as
3011 well as having several new commands. Please see the External IVR page on the Asterisk
3012 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3013 * Added ability to communicate over a TCP socket instead of forking a child process for the
3014 ExternalIVR application.
3015 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3016 of just the first one if you give the function more then one channel to check.
3017 * PrivacyManager now takes an option where you can specify a context where the
3018 given number will be matched. This way you have more control over who is allowed
3019 and it stops the people who blindly enter 10 digits.
3020 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3021 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3022 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3023 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3024 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3025 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3026 * The Dial() application no longer copies the language used by the caller to the callee's
3027 channel. If you desire for the caller's channel's language to be used for file playback
3028 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3029 * SendImage() no longer hangs up the channel on error; instead, it sets the
3030 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3031 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3033 * Park has a new option, 's', which silences the announcement of the parking space number.
3034 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3035 invalid input and will be assumed to mean that no timeout is desired.
3039 * Added DNS manager support to registrations for peers referencing peer entries.
3040 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3041 as well as periodically updating the IP address. These properties allow for
3042 better performance as well as recovery in the event of an IP change.
3043 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3044 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3045 These changes also provide performance improvements for call setup and tear down.
3046 * Added ability to specify registration expiry time on a per registration basis in
3048 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3050 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3051 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3052 * 'sip show peers' and 'sip show users' display their entries sorted in
3053 alphabetical order, as opposed to the order they were in, in the config
3055 * Videosupport now supports an additional option, "always", which always sets
3056 up video RTP ports, even on clients that don't support it. This helps with
3057 callfiles and certain transfers to ensure that if two video phones are
3058 connected, they will always share video feeds.
3062 * Existing DNS manager lookups extended to check for SRV records.
3063 * IAX2 encryption support has been improved to support periodic key rotation
3064 within a call for enhanced security. The option "keyrotate" has been
3065 provided to disable this functionality to preserve backwards compatibility
3066 with older versions of IAX2 that do not support key rotation.
3070 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3071 data tree based on the given <path>.
3072 * New CLI command "data show providers" that will display all the registered
3074 * New CLI command, "config reload <file.conf>" which reloads any module that
3075 references that particular configuration file. Also added "config list"
3076 which shows which configuration files are in use.
3077 * New CLI commands, "pri show version" and "ss7 show version" that will
3078 display which version of libpri and libss7 are being used, respectively.
3079 A new API call was added so trunk will now have to be compiled against
3080 a versions of libpri and libss7 that have them or it will not know that
3081 these libraries exist.
3082 * The commands "core show globals", "core set global" and "core set chanvar" has
3083 been deprecated in favor of the more semanticly correct "dialplan show globals",
3084 "dialplan set chanvar" and "dialplan set global".
3085 * New CLI command "dialplan show chanvar" to list all variables associated
3086 with a given channel.
3090 * Addresses managed by DNS manager now can check to see if there is a DNS
3091 SRV record for a given domain and will use that hostname/port if present.
3093 AMI - The manager (TCP/TLS/HTTP)
3094 --------------------------------
3095 * The Status command now takes an optional list of variables to display
3096 along with channel status.
3097 * The QueueEntry event now also includes the channel's uniqueid
3101 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3102 as some people were running into this limit. This limit has been increased
3107 * The TRANSFER queue log entry now includes the the caller's original
3108 position in the transferred-from queue.
3109 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3110 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3111 as well as an explanation about timeout options in general
3112 * Added a new option - C - for forcing the "answered elsewhere" flag on
3113 cancellation of calls in to members of the queue. This is to avoid the
3114 call to a member of a queue having the call listed as a "missed call".
3118 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3119 adaptive capabilities. What this means in practical terms is that if your
3120 realtime table lacks critical fields, Asterisk will now emit warnings to
3121 that effect. Also, some of the realtime drivers have the ability (if
3122 configured) to automatically add those columns to the table with the
3123 correct type and length.
3127 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3128 the 'setvar' option to cause a given audio file to be played upon completion
3129 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3130 Skinny channels only.
3131 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3132 Hoard page on the Asterisk wiki for more information:
3133 https://wiki.asterisk.org/wiki/x/pQBB
3134 * Config file variables may now be appended to, by using the '+=' append
3135 operator. This is most helpful when working with long SQL queries in
3136 func_odbc.conf, as the queries no longer need to be specified on a single
3138 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3139 which will add a second to the billsec when the ending
3140 time is set, if the number in the microseconds field of the end time is
3141 greater than the number of microseconds in the answer time. This allows
3142 users to count the 'initiated' seconds in their billing records.
3144 ------------------------------------------------------------------------------
3145 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3146 ------------------------------------------------------------------------------
3148 AMI - The manager (TCP/TLS/HTTP)
3149 --------------------------------
3150 * Manager has undergone a lot of changes, all of them documented
3151 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3152 * Manager version has changed to 1.1
3153 * Added a new action 'CoreShowChannels' to list currently defined channels
3154 and some information about them.
3155 * Added a new action 'SIPshowregistry' to list SIP registrations.
3156 * Added TLS support for the manager interface and HTTP server
3157 * Added the URI redirect option for the built-in HTTP server
3158 * The output of CallerID in Manager events is now more consistent.
3159 CallerIDNum is used for number and CallerIDName for name.
3160 * Enable https support for builtin web server.
3161 See configs/http.conf.sample for details.
3162 * Added a new action, GetConfigJSON, which can return the contents of an
3163 Asterisk configuration file in JSON format. This is intended to help
3164 improve the performance of AJAX applications using the manager interface
3166 * SIP and IAX manager events now use "ChannelType" in all cases where we
3167 indicate channel driver. Previously, we used a mixture of "Channel"
3168 and "ChannelDriver" headers.
3169 * Added a "Bridge" action which allows you to bridge any two channels that
3170 are currently active on the system.
3171 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3172 the voicemail users setup.
3173 * Added 'DBDel' and 'DBDelTree' manager commands.
3174 * cdr_manager now reports events via the "cdr" level, separating it from
3175 the very verbose "call" level.
3176 * Manager users are now stored in memory. If you change the manager account
3177 list (delete or add accounts) you need to reload manager.
3178 * Added Masquerade manager event for when a masquerade happens between
3180 * Added "manager reload" command for the CLI
3181 * Lots of commands that only provided information are now allowed under the
3182 Reporting privilege, instead of only under Call or System.
3183 * The IAX* commands now require either System or Reporting privilege, to
3184 mirror the privileges of the SIP* commands.
3185 * Added ability to retrieve list of categories in a config file.
3186 * Added ability to retrieve the content of a particular category.
3187 * Added ability to empty a context.
3188 * Created new action to create a new file.
3189 * Updated delete action to allow deletion by line number with respect to category.
3190 * Added new action insert to add new variable to category at specified line.
3191 * Updated action newcat to allow new category to be inserted in file above another
3193 * Added new event "JitterBufStats" in the IAX2 channel
3194 * Originate now requires the Originate privilege and, if you want to call out
3195 to a subshell, it requires the System privilege, as well. This was done to
3196 enhance manager security.
3197 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3198 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3199 or manager show command Atxfer from the CLI
3200 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3201 details or manager show command IAXregistry from the CLI
3205 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3206 state in the dialplan, as well as creating custom device states that are
3207 controllable from the dialplan.
3208 * Extend CALLERID() function with "pres" and "ton" parameters to
3209 fetch string representation of calling number presentation indicator
3210 and numeric representation of type of calling number value.
3211 * MailboxExists converted to dialplan function
3212 * A new option to Dial() for telling IP phones not to count the call
3213 as "missed" when dial times out and cancels.
3214 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3215 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3216 held for any given channel. Also, locks are automatically freed when a
3218 * Added HINT() dialplan function that allows retrieving hint information.
3219 Hints are mappings between extensions and devices for the sake of
3220 determining the state of an extension. This function can retrieve the list
3221 of devices or the name associated with a hint.
3222 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3224 * Added SYSINFO() dialplan function which allows retrieval of system information
3225 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3226 the existence of a dialplan target.
3227 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3228 upper and lower case, respectively.
3229 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3230 ID for the call (not the Asterisk call ID or unique ID), provided that the
3231 channel driver supports this. For SIP, you get the SIP call-ID for the
3232 bridged channel which you can store in the CDR with a custom field.
3236 * Added CLI permissions, config file: cli_permissions.conf
3237 default is to allow all commands for every local user/group.
3238 Also this new feature added three new CLI commands:
3239 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3240 - cli reload permissions
3241 - cli show permissions
3242 * New CLI command "core show hint" (usage: core show hint <exten>)
3243 * New CLI command "core show settings"
3244 * Added 'core show channels count' CLI command.
3245 * Added the ability to set the core debug and verbose values on a per-file basis.
3246 * Added 'queue pause member' and 'queue unpause member' CLI commands
3247 * Ability to set process limits ("ulimit") without restarting Asterisk
3248 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3249 output to make debugging on busy systems much easier.
3250 * New CLI commands "dialplan set extenpatternmatching true/false"
3251 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3252 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3253 listed in the startup_commands section of cli.conf will get executed.
3254 * Added a CLI command, "devstate change", which allows you to set custom device
3255 states from the func_devstate module that provides the DEVICE_STATE() function
3256 and handling of the "Custom:" devices.
3257 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3258 sorted into the different possible callbacks, with the number of entries
3259 currently scheduled for each. Gives you a feel for how busy the sip channel
3261 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3262 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3263 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3267 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3268 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3269 for a received call. If it is detected, the channel will jump to the
3270 'fax' extension in the dialplan.
3271 * The default SIP useragent= identifier now includes the Asterisk version
3272 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3273 If set, and the incoming request carries authentication info,
3274 the username to match in the users list is taken from the Digest header
3275 rather than from the From: field. This feature is considered experimental.
3276 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3277 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3278 * The "localmask" setting was removed in version 1.2 and the reminder about it
3279 being removed is now also removed.
3280 * A new option "busylevel" for setting a level of calls where asterisk reports
3281 a device as busy, to separate it from call-limit. This value is also added
3282 to the SIP_PEER dialplan function.
3283 * A new realtime family called "sipregs" is now supported to store SIP registration
3284 data. If this family is defined, "sippeers" will be used for configuration and
3285 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3286 registration data, as before.
3287 * The SIPPEER function have new options for port address, call and pickup groups
3288 * Added support for T.140 realtime text in SIP/RTP
3289 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3290 required due to the restructuring of how MWI is handled. See the descriptions
3291 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3292 for more information.
3293 * Added rtpdest option to CHANNEL() dialplan function.
3294 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3295 * SIP now adds a header to the CANCEL if the call was answered by another phone
3296 in the same dial command, or if the new c option in dial() is used.
3297 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3298 states it is not needed. For phones, however, that do require it the "registertrying" option
3299 has been added so it can be enabled.
3300 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3301 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3302 used to enable this functionality).
3303 * New settings for timer T1 and timer B on a global level or per device. This makes it
3304 possible to force timeout faster on non-responsive SIP servers. These settings are
3305 considered advanced, so don't use them unless you have a problem.
3306 * Added a dial string option to be able to set the To: header in an INVITE to any
3308 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3309 the qualify frequency.
3310 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3311 were not properly torn down due to network or endpoint failures during an established
3313 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3314 and configs/sip.conf.sample for more information on how it is used.
3315 * Added a new configuration option "authfailureevents" that enables manager events when
3316 a peer can't authenticate properly.
3317 * Added DNS manager support to registrations for peers not referencing a peer entry.
3321 * Added the trunkmaxsize configuration option to chan_iax2.
3322 * Added the srvlookup option to iax.conf
3323 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3326 XMPP Google Talk/Jingle changes
3327 -------------------------------
3328 * Added the bindaddr option to gtalk.conf.
3332 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
3333 * Proper codec support in chan_skinny.
3334 * Added settings for IP and Ethernet QoS requests
3338 * Added separate settings for media QoS in mgcp.conf
3340 Console Channel Driver changes
3341 ------------------------------
3342 * Added experimental support for video send & receive to chan_oss.
3343 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
3346 Phone channel changes (chan_phone)
3347 ----------------------------------
3348 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
3350 H.323 channel Changes
3351 ---------------------
3352 * H323 remote hold notification support added (by NOTIFY message
3353 and/or H.450 supplementary service)
3355 Local channel changes
3356 ---------------------
3357 * The device state functionality in the Local channel driver has been updated
3358 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
3359 to just UNKNOWN if the extension exists.
3360 * Added jitterbuffer support for chan_local. This allows you to use the
3361 generic jitterbuffer on incoming calls going to Asterisk applications.
3362 For example, this would allow you to use a jitterbuffer for an incoming
3363 SIP call to Voicemail by putting a Local channel in the middle. This
3364 feature is enabled by using the 'j' option in the Dial string to the Local
3365 channel in conjunction with the existing 'n' option for local channels.
3366 * A 'b' option has been added which causes chan_local to return the actual channel
3367 that is behind it when queried. This is useful for transfer scenarios as the
3368 actual channel will be transferred, not the Local channel.
3370 Agent channel changes
3371 ----------------------
3372 * The ackcall and endcall options are now supplemented with options acceptdtmf
3373 and enddtmf. These allow for the DTMF keypress to be configurable. The options
3374 default to their old hard-coded values ('#' and '*' respectively) so this should
3375 not break any existing agent installations.
3377 DAHDI channel driver (chan_dahdi) Changes
3378 ----------------------------------------
3379 * SS7 support (via libss7 library)
3380 * In India, some carriers transmit CID via dtmf. Some code has been added
3381 that will handle some situations. The cidstart=polarity_IN choice has been added for
3382 those carriers that transmit CID via dtmf after a polarity change.
3383 * CID matching information is now shown when doing 'dialplan show'.
3384 * Added dahdi show version CLI command.
3385 * Added setvar support to chan_dahdi.conf channel entries.
3386 * Added two new options: mwimonitor and mwimonitornotify. These options allow
3387 you to enable MWI monitoring on FXO lines. When the MWI state changes,
3388 the script specified in the mwimonitornotify option is executed. An internal
3389 event indicating the new state of the mailbox is also generated, so that
3390 the normal MWI facilities in Asterisk work as usual.
3391 * Added signalling type 'auto', which attempts to use the same signalling type
3392 for a channel as configured in DAHDI. This is primarily designed for analog
3393 ports, but will also work for digital ports that are configured for FXS or FXO
3394 signalling types. This mode is also the default now, so if your chan_dahdi.conf
3395 does not specify signalling for a channel (which is unlikely as the sample
3396 configuration file has always recommended specifying it for every channel) then
3397 the 'auto' mode will be used for that channel if possible.
3398 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
3399 state for a channel; also ensured that the DNDState Manager event is
3400 emitted no matter how the DND state is set or cleared.
3404 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
3405 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
3406 for details. This new channel driver allows you to use Nortel i2002,
3407 i2004, and i2050 phones with Asterisk.
3408 * Added a new channel driver, chan_console, which uses portaudio as a cross
3409 platform audio interface. It was written as a channel driver that would
3410 work with Mac CoreAudio, but portaudio supports a number of other audio
3411 interfaces, as well. Note that this channel driver requires v19 or higher
3412 of portaudio; older versions have a different API.
3416 * Added the ability to specify arguments to the Dial application when using
3417 the DUNDi switch in the dialplan.
3418 * Added the ability to set weights for responses dynamically. This can be
3419 done using a global variable or a dialplan function. Using the SHELL()
3420 function would allow you to have an external script set the weight for
3422 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
3423 functions will allow you to initiate a DUNDi query from the dialplan,
3424 find out how many results there are, and access each one.
3425 * Added the ability to specifiy a port for a dundi peer.
3429 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
3430 functions will allow you to initiate an ENUM lookup from the dialplan,
3431 and Asterisk will cache the results. ENUMRESULT can be used to access
3432 the results without doing multiple DNS queries.
3436 * Added the ability to customize which sound files are used for some of the
3437 prompts within the Voicemail application by changing them in voicemail.conf
3438 * Added the ability for the "voicemail show users" CLI command to show users
3439 configured by the dynamic realtime configuration method.
3440 * MWI (Message Waiting Indication) handling has been significantly
3441 restructured internally to Asterisk. It is now totally event based
3442 instead of polling based. The voicemail application will notify other
3443 modules that have subscribed to MWI events when something in the mailbox
3445 This also means that if any other entity outside of Asterisk is changing
3446 the contents of mailboxes, then the voicemail application still needs to
3447 poll for changes. Examples of situations that would require this option
3448 are web interfaces to voicemail or an email client in the case of using
3449 IMAP storage. So, two new options have been added to voicemail.conf
3450 to account for this: "pollmailboxes" and "pollfreq". See the sample
3451 configuration file for details.
3452 * Added "tw" language support
3453 * Added support for storage of greetings using an IMAP server
3454 * Added ability to customize forward, reverse, stop, and pause keys for message playback
3455 * SMDI is now enabled in voicemail using the smdienable option.
3456 * A "lockmode" option has been added to asterisk.conf to configure the file
3457 locking method used for voicemail, and potentially other things in the
3458 future. The default is the old behavior, lockfile. However, there is a
3459 new method, "flock", that uses a different method for situations where the
3460 lockfile will not work, such as on SMB/CIFS mounts.
3461 * Added the ability to backup deleted messages, to ease recovery in the case
3462 that a user accidentally deletes a message, and discovers that they need it.
3463 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
3464 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
3465 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
3466 voicemail boxes. The SMDI interface can also poll for MWI changes when some
3467 outside entity is modifying the state of the mailbox (such as IMAP storage or
3468 a web interface of some kind).
3469 * Added the support for marking messages as "urgent." There are two methods to accomplish
3470 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
3471 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
3472 the message as urgent after he has recorded a voicemail by following the voice instructions.
3473 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
3478 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
3479 used across multiple queues.
3480 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
3481 setqueueentryvar options for each queue, see queues.conf.sample for details.
3482 * Added keepstats option to queues.conf which will keep queue
3483 statistics during a reload.
3484 * setinterfacevar option in queues.conf also now sets a variable
3485 called MEMBERNAME which contains the member's name.
3486 * Added 'Strategy' field to manager event QueueParams which represents
3487 the queue strategy in use.
3488 * Added option to run macro when a queue member is connected to a caller,
3489 see queues.conf.sample for details.
3490 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
3491 does not count paused queue members as unavailable.
3492 * Added min-announce-frequency option to queues.conf which allows you to control the
3493 minimum amount of time between queue announcements for use when the caller's queue
3494 position changes frequently.
3495 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
3497 * Added ability for non-realtime queues to have realtime members
3498 * Added the "linear" strategy to queues.
3499 * Added the "wrandom" strategy to queues.
3500 * Added new channel variable QUEUE_MIN_PENALTY
3501 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
3502 rules in queuerules.conf. See configs/queuerules.conf.sample for details
3503 * Added a new parameter for member definition, called state_interface. This may be
3504 used so that a member may be called via one interface but have a different interface's
3505 device state reported.
3506 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
3507 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
3508 "manager show command QueueReset."
3509 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
3510 specified by the periodic-announce option, then one will be chosen randomly when it is time
3511 to play a periodic announcment
3512 * New configuration options: announce-position now takes two more values in addition to "yes" and
3513 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
3514 announce-position-limit. By setting announce-position to "limit" callers will only have their
3515 position announced if their position is less than what is specified by announce-position-limit.
3516 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
3517 will be told that their are more than announce-position-limit callers waiting.
3518 * Two new queue log events have been added. An ADDMEMBER event will be logged
3519 when a realtime queue member is added and a REMOVEMEMBER event will be logged
3520 when a realtime queue member is removed. Since there is no calling channel associated
3521 with these events, the string "REALTIME" is placed where the channel's unique id
3522 is typically placed.
3523 * The configuration method for the "joinempty" and "leavewhenempty" options has
3524 changed to a comma-separated list of methods of determining member availability
3525 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
3526 values are still accepted for backwards-compatibility, though.
3527 * The average talktime is now calculated on queues. This information is reported via the
3528 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
3529 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
3534 * The 'o' option to provide an optimization has been removed and its functionality
3535 has been enabled by default.
3536 * When a conference is created, the UNIQUEID of the channel that caused it to be
3537 created is stored. Then, every channel that joins the conference will have the
3538 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
3539 callers that come and go from long standing conferences.
3540 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
3541 except it does operations on a channel by name, instead of number in a conference.
3542 This is a very useful feature in combination with the 'X' option to ChanSpy.
3543 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
3545 * Added new RealTime functionality to provide support for scheduled conferencing.
3546 This includes optional messages to the caller if they attempt to join before
3547 the schedule start time, or to allow the caller to join the conference early.
3548 Also included is optional support for limiting the number of callers per
3549 RealTime conference.
3550 * Added the S() and L() options to the MeetMe application. These are pretty
3551 much identical to the S() and L() options to Dial(). They let you set
3552 timeouts for the conference, as well as have warning sounds played to
3553 let the caller know how much time is left, and when it is running out.
3554 * Added the ability to do "meetme concise" with the "meetme" CLI command.
3555 This extends the concise capabilities of this CLI command to include
3556 listing all conferences, instead of an addition to the other sub commands
3557 for the "meetme" command.
3558 * Added the ability to specify the music on hold class used to play into the
3559 conference when there is only one member and the M option is used.
3560 * Added MEETME_INFO dialplan function which provides a way to query
3561 various properties of a Meetme conference.
3562 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
3563 and *84: record in-conf
3565 Other Dialplan Application Changes
3566 ----------------------------------
3567 * Argument support for Gosub application
3568 * From the to-do lists: straighten out the app timeout args:
3569 Wait() app now really does 0.3 seconds- was truncating arg to an int.
3570 WaitExten() same as Wait().
3571 Congestion() - Now takes floating pt. argument.
3572 Busy() - now takes floating pt. argument.
3573 Read() - timeout now can be floating pt.
3574 WaitForRing() now takes floating pt timeout arg.
3575 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
3576 * Added 's' option to Page application.
3577 * Added an optional timeout argument to the Page application.
3578 * Added 'E', 'V', and 'P' commands to ExternalIVR.
3579 * Added 'o' and 'X' options to Chanspy.
3580 * Added a new dialplan application, Bridge, which allows you to bridge the
3581 calling channel to any other active channel on the system.
3582 * Added the ability to specify a music on hold class to play instead of ringing
3583 for the SLATrunk application.
3584 * The Read application no longer exits the dialplan on error. Instead, it sets
3585 READSTATUS to ERROR, which you can catch and handle separately.
3586 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
3587 of asking for verification of each name, one at a time.
3588 * Privacy() no longer uses privacy.conf, as all options are specifyable as
3589 direct options to the app.
3590 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
3592 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
3593 * The ChannelRedirect application no longer exits the dialplan if the given channel
3594 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
3595 or NOCHANNEL if the given channel was not found.
3596 * The silencethreshold setting that was previously configurable in multiple
3597 applications is now settable globally via dsp.conf.
3599 Music On Hold Changes
3600 ---------------------
3601 * A new option, "digit", has been added for music on hold classes in
3602 musiconhold.conf. If this is set for a music on hold class, a caller
3603 listening to music on hold can press this digit to switch to listening
3604 to this music on hold class.
3605 * Support for realtime music on hold has been added.
3606 * In conjunction with the realtime music on hold, a general section has
3607 been added to musiconhold.conf, its sole variable is cachertclasses. If this
3608 is set, then music on hold classes found in realtime will be cached in memory.
3612 * AEL upgraded to use the Gosub with Arguments instead
3613 of Macro application, to hopefully reduce the problems
3614 seen with the artificially low stack ceiling that
3615 Macro bumps into. Macros can only call other Macros
3616 to a depth of 7. Tests run using gosub, show depths
3617 limited only by virtual memory. A small test demonstrated
3618 recursive call depths of 100,000 without problems.
3619 -- in addition to this, all apps that allowed a macro
3620 to be called, as in Dial, queues, etc, are now allowing
3621 a gosub call in similar fashion.
3622 * AEL now generates LOCAL(argname) declarations when it
3623 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
3624 etc. That makes the arguments local in scope. The user
3625 can define their own local variables in macros, now,
3626 by saying "local myvar=someval;" or using Set() in this
3627 fashion: Set(LOCAL(myvar)=someval); ("local" is now
3629 * utils/conf2ael introduced. Will convert an extensions.conf
3630 file into extensions.ael. Very crude and unfinished, but
3631 will be improved as time goes by. Should be useful for a
3632 first pass at conversion.
3633 * aelparse will now read extensions.conf to see if a referenced
3634 macro or context is there before issueing a warning.
3635 * AEL parser sets a local channel variable ~~EXTEN~~, to
3636 preserve the value of ${EXTEN} thru switch statements.
3637 * New operator in $[...] expressions: the ~~ operator serves
3638 as a concatenation operator. AT THE MOMENT, it is really only
3639 necessary and useful in AEL, especially in if() expressions.
3640 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
3641 any enclosing double-quotes, and evaluate to the value of a
3642 concatenated with the value of b. For example if a is set to
3643 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
3644 evaluate to xyzabc .
3647 Call Features (res_features) Changes
3648 ------------------------------------
3649 * Added the parkedcalltransfers option to features.conf
3650 * Added parkedcallparking option to control one touch parking w/ parking
3652 * Added parkedcallhangup option to control disconnect feature w/ parking
3654 * Added parkedcallrecording option to control one-touch record w/ parking
3656 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
3657 parkedcalltransfers option support for multiple parking lots.
3658 * Added BRIDGE_FEATURES variable to set available features for a channel
3659 * The built-in method for doing attended transfers has been updated to
3660 include some new options that allow you to have the transferee sent
3661 back to the person that did the transfer if the transfer is not successful.
3662 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
3663 in features.conf.sample.
3664 * Added support for configuring named groups of custom call features in
3665 features.conf. This means that features can be written a single time, and
3666 then mapped into groups of features for different key mappings or easier
3668 * Updated the ParkedCall application to allow you to not specify a parking
3669 extension. If you don't specify a parking space to pick up, it will grab
3670 the first one available.
3671 * Added cli command 'features reload' to reload call features from features.conf
3672 * Moved into core asterisk binary.
3673 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
3674 * Added the ability for custom parking lots to be configured with their own
3675 parking extension with the parkext option.
3677 Language Support Changes
3678 ------------------------
3679 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
3680 * Added support for the Hungarian language for saying numbers, dates, and times.
3684 * Added SPEECH commands for speech recognition. A complete listing can be found
3686 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
3687 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
3688 does not behave as expected; the native command needs to be used, instead.
3689 * Added the ability to perform SRV lookups on fast AGI calls. To use this
3690 feature, simply use hagi: instead of agi: as the protocol portion
3691 of the URI parameter to the AGI function call in your dial plan. Also note
3692 that specifying a port number in the AGI URI will disable SRV lookups,
3693 even if you use the hagi: protocol.
3694 * No longer support MSG_OOB flag on HANGUP.
3698 * Added rotatestrategy option to logger.conf, along with two new options:
3699 "timestamp" which will use the time to name the logger files instead of
3700 sequence number; and "rotate", which rotates the names of the log files,
3701 similar to the way syslog rotates files.
3702 * Added exec_after_rotate option to logger.conf, which allows a system
3703 command to be run after rotation. This is primarily useful with
3704 rotatestrategy=rotate, to allow a limit on the number of log files kept
3705 and to ensure that the oldest log file gets deleted.
3706 * Added realtime support for the queue log
3710 * The cdr_manager module has a [mappings] feature, like cdr_custom,
3711 to add fields to the manager event from the CDR variables.
3712 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
3713 backend database CDR table. Specifically, additional, non-standard
3714 columns are supported, merely by setting the corresponding CDR variable in
3715 your dialplan. In addition, you may alias any column to another name (for
3716 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
3717 simply "alias src => ANI" in the configuration file). Records may be
3718 posted to more than one backend, simply by specifying multiple categories
3719 in the configuration file. And finally, you may filter which CDRs get
3720 posted to each backend, by specifying a filter (which the record must
3721 match) for the particular category. Filters are additive (meaning all
3722 rules must match to post that CDR).
3723 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
3724 module. Specifically, you may add additional columns into the table and
3725 they will be set, if you set the corresponding CDR variable name. Also,
3726 if you omit columns in your database table, they will be silently skipped
3727 (but a record will still be inserted, based on what columns remain). Note
3728 that the other two features from cdr_adaptive_odbc (alias and filter) are
3729 not currently supported.
3730 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
3731 has been disabled using the NoCDR application.
3733 Miscellaneous New Modules
3734 -------------------------
3735 * Added a new CDR module, cdr_sqlite3_custom.
3736 * Added a new realtime configuration module, res_config_sqlite
3737 * Added a new codec translation module, codec_resample, which re-samples
3738 signed linear audio between 8 kHz and 16 kHz to help support wideband
3740 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
3741 based on configuration templates that use Asterisk dialplan function and
3742 variable substitution. It should be possible to create phone profiles and
3743 templates that work for the majority of phones provisioned over http. It
3744 is currently only intended to provision a single user account per phone.
3745 An example profile and set of templates for Polycom phones is provided.
3746 NOTE: Polycom firmware is not included, but should be placed in
3747 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
3748 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
3749 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
3750 provided; there is a JACK() application, and a JACK_HOOK() function. Both
3751 interfaces create an input and output JACK port. The application makes
3752 these ports the endpoint of the call. The audio coming from the channel
3753 goes out the output port and whatever comes back in on the input port is
3754 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
3755 audiohook on the channel. This lets you run the audio coming from a
3756 channel through JACK, and whatever comes back in is what gets forwarded
3757 on as the channel's audio. This is very useful for building custom
3758 vocoders or doing recording or analysis of the channel's audio in another
3760 * Added a new module, res_config_curl, which permits using a HTTP POST url
3761 to retrieve, create, update, and delete realtime information from a remote
3762 web server. Note that this module requires func_curl.so to be loaded for
3763 backend functionality.
3764 * Added a new module, res_config_ldap, which permits the use of an LDAP
3765 server for realtime data access.
3766 * Added support for writing and running your dialplan in lua using the pbx_lua
3767 module. See configs/extensions.lua.sample for examples of how to do this.
3771 * Ability to use libcap to set high ToS bits when non-root
3772 on Linux. If configure is unable to find libcap then you
3773 can use --with-cap to specify the path.
3774 * Added maxfiles option to options section of asterisk.conf which allows you to specify
3775 what Asterisk should set as the maximum number of open files when it loads.
3776 * Added the jittertargetextra configuration option.
3777 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
3778 configuration files for the IP channel drivers. The new option is "cos".
3779 This information is also documented on the Asterisk wiki at
3780 https://wiki.asterisk.org/wiki/x/EYBG
3781 * When originating a call using AMI or pbx_spool that fails the reason for failure
3782 will now be available in the failed extension using the REASON dialplan variable.
3783 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
3784 It allows you to configure a prefix for auto-monitor recordings.
3785 * A new extension pattern matching algorithm, based on a trie, is introduced
3786 here, that could noticeably speed up mid-sized to large dialplans.
3787 It is NOT used by default, as duplicating the behaviour of the old pattern
3788 matcher is still under development. A config file option, in extensions.conf,
3789 in the [general] section, called "extenpatternmatchingnew", is by default
3790 set to false; setting that to true will force the use of the new algorithm.
3791 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
3792 be used to switch the algorithms at run time.
3793 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
3794 specifying which socket to use to connect to the running Asterisk daemon
3796 * Performance enhancements to the sched facility, which is used in
3797 the channel drivers, etc. Added hashtabs and doubly-linked lists
3798 to speed up deletion; start at the beginning or end of list to
3800 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
3801 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
3802 Added regression tests to the tests/ dir, also.
3803 * Added a refcount trace feature to astobj2 for those trying to balance
3804 object creation, deletion; work, play; space and time. See the
3805 notes in astobj2.h. Also, see utils/refcounter as well, as a
3806 quick way to find unbalanced refcounts in what could be a sea
3807 of objects that were balanced.
3808 * Added logging to 'make update' command. See update.log
3809 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
3810 do not come from the remote party.
3811 * Added the 'n' option to the SpeechBackground application to tell it to not
3812 answer the channel if it has not already been answered.
3813 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
3814 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
3816 * iLBC source code no longer included (see UPGRADE.txt for details)
3817 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
3818 deadlock is detected, a backtrace of the stack which led to the lock calls
3819 will be output to the CLI.
3820 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
3821 the "core show locks" CLI command will give lock information output as well
3822 as a backtrace of the stack which led to the lock calls.
3823 * users.conf now sports an optional alternateexts property, which permits
3824 allocation of additional extensions which will reach the specified user.
3825 * A new option for the configure script, --enable-internal-poll, has been added
3826 for use with systems which may have a buggy implementation of the poll system
3827 call. If you notice odd behavior such as the CLI being unresponsive on remote
3828 consoles, you may want to try using this option. This option is enabled by default
3829 on Darwin systems since it is known that the Darwin poll() implementation has
3833 --------------------
3834 * In addition to timing from DAHDI, there is a new timing module called
3835 res_timing_timerfd. In order to use this, you must be running Linux with
3836 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
3837 script will be able to tell if you have the requirements. From menuselect, select
3838 res_timing_timerfd from the Resource Modules menu.
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