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 * Record application now has an option 'o' which allows 0 to act as an exit
18 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
21 --------------------------
22 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
23 as the chanprefix parameter if the 'u' option is specified.
26 --------------------------
27 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
28 conference user menus.
30 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
31 menus, bridge settings, and user settings that have been applied by the
32 CONFBRIDGE dialplan function.
34 * The ConfBridge dialplan application now sets a channel variable,
35 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
36 how a channel exited the conference.
38 * Added conference user option 'announce_join_leave_review'. This option
39 implies 'announce_join_leave' with the added effect that the user will
40 be asked if they want to confirm or re-record the recording of their
41 name when entering the conference
44 --------------------------
45 * At exit, the Directory application now sets a channel variable
46 DIRECTORY_RESULT to one of the following based on the reason for exiting:
47 OPERATOR user requested operator by pressing '0' for operator
48 ASSISTANT user requested assistant by pressing '*' for assistant
49 TIMEOUT user pressed nothing and Directory stopped waiting
50 HANGUP user's channel hung up
51 SELECTED user selected a user from the directory and is routed
52 USEREXIT user pressed '#' from the selection prompt to exit
53 FAILED directory failed in a way that wasn't accounted for. Dang.
56 --------------------------
57 * MusicOnHold streams (all modes other than "files") now support wide band
61 --------------------------
62 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
63 and for the channel executing Page respectively.
66 --------------------------
67 * PickupChan now accepts channel uniqueids of channels to pickup.
70 --------------------------
71 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
72 to 'true' (case insensitive), then any Say application (SayNumber,
73 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
74 anticipate DTMF. If DTMF is received, these applications will behave like
75 the background application and jump to the received extension once a match
76 is established or after a short period of inactivity.
79 -------------------------
80 * A new function, MIXMONITOR, has been added to allow access to individual
81 instances of MixMonitor on a channel.
84 -------------------------
85 * Core Show Locks output now includes Thread/LWP ID if the platform
86 supports this feature.
88 ------------------------------------------------------------------------------
89 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
90 ------------------------------------------------------------------------------
94 * Added a new module that provides AMI control over MWI within Asterisk,
95 res_mwi_external_ami. Note that this module depends on res_mwi_external;
96 for more information on enabling this module, see res_mwi_external.
97 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
98 the MWIGet/MWIGetComplete events.
100 * The DialStatus field in the DialEnd event can now contain additional
101 statuses that convey how the dial operation terminated. This includes
102 ABORT, CONTINUE, and GOTO.
104 * Bridge related events now have two additional fields: BridgeName and
105 BridgeCreator. BridgeName is a descriptive name for the bridge;
106 BridgeCreator is the name of the entity that created the bridge. This
107 affects the following events: ConfbridgeStart, ConfbridgeEnd,
108 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
109 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
110 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
114 * The Bridge data model now contains the additional fields 'name' and
115 'creator'. The 'name' field conveys a descriptive name for the bridge;
116 the 'creator' field conveys the name of the entity that created the bridge.
117 This affects all responses to HTTP requests that return a Bridge data model
118 as well as all event derived data models that contain a Bridge data model.
119 The POST /bridges operation may now optionally specify a name to give to
120 the bridge being created.
122 * Added a new ARI resource 'mailboxes' which allows the creation and
123 modification of mailboxes managed by external MWI. Modules res_mwi_external
124 and res_stasis_mailbox must be enabled to use this resource. For more
125 information on external MWI control, see res_mwi_external.
127 * Added new events for externally initiated transfers. The event
128 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
129 of a bridge in the ARI controlled application to the dialplan; the
130 BridgeAttendedTransfer event is raised when a channel initiates an
131 attended transfer of a bridge in the ARI controlled application to the
134 * Channel variables may now be specified as a body parameter to the
135 POST /channels operation. The 'variables' key in the JSON is interpreted
136 as a sequence of key/value pairs that will be added to the created channel
137 as channel variables. Other parameters in the JSON body are treated as
138 query parameters of the same name.
142 * Path support has been added with the 'support_path' option in registration
145 * A 'debug' option has been added to the globals section that will allow
146 sip messages to be logged.
148 * A 'set_var' option has been added to endpoints that will automatically
149 set the desired variable(s) on a channel created for that endpoint.
151 * Several new tables and columns have been added to the realtime schema for
152 the res_pjsip related modules. See the UPGRADE.txt notes for updating
157 * A new module, res_mwi_external, has been added to Asterisk. This module
158 acts as a base framework that other modules can build on top of to allow
159 an external system to control MWI within Asterisk. For implementations
160 that make use of res_mwi_external, see res_mwi_external_ami and
161 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
162 that may produce MWI themselves, such as app_voicemail. res_mwi_external
163 and other modules that depend on it cannot be built or loaded with
164 app_voicemail present.
167 ------------------------------------------------------------------------------
168 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
169 ------------------------------------------------------------------------------
174 Asterisk 12 is a standard release of the Asterisk project. As such, the
175 focus of development for this release was on core architectural changes and
176 major new features. This includes:
177 * A more flexible bridging core based on the Bridging API
178 * A new internal message bus, Stasis
179 * Major standardization and consistency improvements to AMI
180 * Addition of the Asterisk RESTful Interface (ARI)
181 * A new SIP channel driver, chan_pjsip
182 In addition, as the vast majority of bridging in Asterisk was migrated to the
183 Bridging API used by ConfBridge, major changes were made to most of the
184 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
186 Specifications have been written for the affected interfaces. These
187 specifications are available on the Asterisk wiki:
188 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
189 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
190 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
192 It is *highly* recommended that anyone migrating to Asterisk 12 read the
193 information regarding its release both in this file and in the accompanying
194 UPGRADE.txt file. More detailed information on the major changes can be found
195 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
200 * Added build option DISABLE_INLINE. This option can be used to work around a
201 bug in gcc. For more information, see
202 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
204 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
205 the CHANNEL_TRACE build option were incompatible with the new bridging
208 * Asterisk now optionally uses libxslt to improve XML documentation generation
209 and maintainability. If libxslt is not available on the system, some XML
210 documentation will be incomplete.
212 * Asterisk now depends on libjansson. If a package of libjansson is not
213 available on your distro, please see http://www.digip.org/jansson/.
215 * Asterisk now depends on libuuid and, optionally, uriparser. It is
216 recommended that you install uriparser, even if it is optional.
218 * The new SIP stack and channel driver uses a particular version of PJSIP.
219 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
220 configuring and installing PJSIP for usage with Asterisk.
222 * Optional API was re-implemented to be more portable, and no longer requires
223 weak reference support from the compiler. The build option OPTIONAL_API may
224 be disabled to disable Optional API support.
231 * Along with AgentRequest, this application has been modified to be a
232 replacement for chan_agent. The act of a channel calling the AgentLogin
233 application places the channel into a pool of agents that can be
234 requested by the AgentRequest application. Note that this application, as
235 well as all other agent related functionality, is now provided by the
236 app_agent_pool module. See chan_agent and AgentRequest for more information.
238 * This application no longer performs agent authentication. If authentication
239 is desired, the dialplan needs to perform this function using the
240 Authenticate or VMAuthenticate application or through an AGI script before
243 * If this application is called and the agent is already logged in, the
244 dialplan will continue exection with the AGENT_STATUS channel variable set
245 to ALREADY_LOGGED_IN.
247 * The agents.conf schema has changed. Rather than specifying agents on a
248 single line in comma delineated fashion, each agent is defined in a separate
249 context. This allows agents to use the power of context templates in their
252 * A number of parameters from agents.conf have been removed. This includes
253 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
254 urlprefix, and savecallsin. These options were obsoleted by the move from
255 a channel driver model to the bridging/application model provided by
260 * A new application, this will request a logged in agent from the pool and
261 bridge the requested channel with the channel calling this application.
262 Logged in agents are those channels that called the AgentLogin application.
263 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
264 application will be set with an appropriate error value.
268 * This application has been removed. It was a holdover from when
269 AgentCallbackLogin was removed.
273 * Added support for additional Ademco DTMF signalling formats, including
274 Express 4+1, Express 4+2, High Speed and Super Fast.
276 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
277 call time, in milliseconds, to run the application.
279 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
280 maximum number of times to retry the call.
282 * Added a new configuration option answait. If set, the AlarmReceiver
283 application will wait the number of milliseconds specified by answait
284 after the channel has answered. Valid values range between 500
285 milliseconds and 10000 milliseconds.
287 * Added configuration option no_group_meta. If enabled, grouping of metadata
288 information in the AlarmReceiver log file will be skipped.
292 * It is now no longer possible to bypass updating the CDR on the channel
293 when answering. CDRs reflect the state of the channel and will always
294 reflect the time they were Answered.
298 * A new application in Asterisk, this will place the calling channel
299 into a holding bridge, optionally entertaining them with some form of
300 media. Channels participating in a holding bridge do not interact with
301 other channels in the same holding bridge. Optionally, however, a channel
302 may join as an announcer. Any media passed from an announcer channel is
303 played to all channels in the holding bridge. Channels leave a holding
304 bridge either when an optional timer expires, or via the ChannelRedirect
305 application or AMI Redirect action.
309 * All participants in a bridge can now be kicked out of a conference room
310 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
311 command, i.e., 'confbridge kick <conference> all'
313 * CLI output for the 'confbridge list' command has been improved. When
314 displaying information about a particular bridge, flags will now be shown
315 for the participating users indicating properties of that user.
317 * The ConfbridgeList event now contains the following fields: WaitMarked,
318 EndMarked, and Waiting. This displays additional properties about the
319 user's profile, as well as whether or not the user is waiting for a
320 Marked user to enter the conference.
322 * Added a new option for conference recording, record_file_append. If enabled,
323 when the recording is stopped and then re-started, the existing recording
324 will be used and appended to.
326 * ConfBridge now has the ability to set the language of announcements to the
327 conference. The language can be set on a bridge profile in confbridge.conf
328 or by the dialplan function CONFBRIDGE(bridge,language)=en.
332 * The channel variable CPLAYBACKSTATUS may now return the value
333 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
334 such as AMI. See the AMI action ControlPlayback for more information.
338 * Added the 'a' option, which allows the caller to enter in an additional
339 alias for the user in the directory. This option must be used in conjunction
340 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
341 specified in voicemail.conf.
345 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
346 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
347 containing the unique ID of the bridge that the channel happens to be in.
351 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
352 for more information.
354 * Variables are no longer purged from the original CDR. See the 'v' option for
357 * The 'A' option has been removed. The Answer time on a CDR is never updated
360 * The 'd' option has been removed. The disposition on a CDR is a function of
361 the state of the channel and cannot be altered.
363 * The 'D' option has been removed. Who the Party B is on a CDR is a function
364 of the state of the respective channels involved in the CDR and cannot be
367 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
368 such that the start time and, if applicable, the answer time was updated.
369 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
370 'r' option now triggers the Reset, setting the start time (and answer time
371 if applicable) to the current time. Note that the 'a' option still sets
372 the answer time to the current time if the channel was already answered.
374 * The 's' option has been removed. A variable can be set on the original CDR
375 if desired using the CDR function, and removed from a forked CDR using the
378 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
379 longer applies in the CDR engine.
381 * The 'v' option now prevents the copy of the variables from the original CDR
382 to the forked CDR. Previously the variables were always copied but were
383 removed from the original. This was changed as removing variables from a CDR
384 can have unintended side effects - this option allows the user to prevent
385 propagation of variables from the original to the forked without modifying
390 * Added the 'n' option to MeetMe to prevent application of the DENOISE
391 function to a channel joining a conference. Some channel drivers that vary
392 the number of audio samples in a voice frame will experience significant
393 quality problems if a denoiser is attached to the channel; this option gives
394 them the ability to remove the denoiser without having to unload func_speex.
398 * The 'b' option now includes conferences as well as sounds played to the
401 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
402 running during a transfer. If a MixMonitor is started on a channel,
403 the MixMonitor will continue to record the audio passing through the
404 channel even in the presence of transfers.
408 * The NoCDR application is deprecated. Please use the CDR_PROP function to
411 * While the NoCDR application will prevent CDRs for a channel from being
412 propagated to registered CDR backends, it will not prevent that data from
413 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
414 function that enables CDRs on a channel will restore those records that have
415 not yet been finalized.
419 * The app_parkandannounce module has been removed. The application
420 ParkAndAnnounce is now provided by the res_parking module. See the
421 res_parking changes for more information.
425 * Added queue available hint. The hint can be added to the dialplan using the
426 following syntax: exten,hint,Queue:{queue_name}_avail
427 For example, if the name of the queue is 'markq':
428 exten => 8501,hint,Queue:markq_avail
429 This will report 'InUse' if there are no logged in agents or no free agents.
430 It will report 'Idle' when an agent is free.
432 * Queues now support a hint for member paused state. The hint uses the form
433 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
434 are the name of the queue and the name of the member to subscribe to,
435 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
436 Members will show as In Use when paused.
438 * The configuration options eventwhencalled and eventmemberstatus have been
439 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
440 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
441 sent. The "Variable" fields will also no longer exist on the Agent* events.
442 These events can be filtered out from a connected AMI client using the
443 eventfilter setting in manager.conf.
445 * The queue log now differentiates between blind and attended transfers. A
446 blind transfer will result in a BLINDTRANSFER message with the destination
447 context and extension. An attended transfer will result in an
448 ATTENDEDTRANSFER message. This message will indicate the method by which
449 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
450 for running an application on a bridge or channel, or "LINK" for linking
451 two bridges together with local channels. The queue log will also now detect
452 externally initiated blind and attended transfers and record the transfer
455 * When performing queue pause/unpause on an interface without specifying an
456 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
457 least one member of any queue exists for that interface.
459 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
460 for realtime queue log entries.
464 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
465 CDRs when they were previously disabled on a channel.
467 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
468 backends occurs on an as-needed basis in order to preserve linkedid
469 propagation and other needed behavior.
473 * A new application, this is similar to SayAlpha except that it supports
474 case sensitive playback of the specified characters. For example,
475 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
479 * This application is deprecated in favor of CHANNEL(amaflags).
483 * The SendDTMF application will now accept 'W' as valid input. This will cause
484 the application to delay one second while streaming DTMF.
488 * A new application in Asterisk 12, this hands control of the channel calling
489 the application over to an external system. Currently, external systems
490 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
494 * UserEvent will now handle duplicate keys by overwriting the previous value
497 * In addition to AMI, UserEvent invocations will now be distributed to any
498 interested Stasis applications.
502 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
503 system as mailbox@context. The rest of the system cannot add @default
504 to mailbox identifiers for app_voicemail that do not specify a context
505 any longer. It is a mailbox identifier format that should only be
506 interpreted by app_voicemail.
508 * The voicemail.conf configuration file now has an 'alias' configuration
509 parameter for use with the Directory application. The voicemail realtime
510 database table schema has also been updated with an 'alias' column.
515 * Pass through support has been added for both VP8 and Opus.
517 * Added format attribute negotiation for the Opus codec. Format attribute
518 negotiation is provided by the res_format_attr_opus module.
523 * Masquerades as an operation inside Asterisk have been effectively hidden
524 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
525 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
526 dropping of frame/audio hooks, and other internal implementation details
527 that users had to deal with. This fundamental change has large implications
528 throughout the changes documented for this version. For more information
529 about the new core architecture of Asterisk, please see the Asterisk wiki.
531 * Multiple parties in a bridge may now be transferred. If a participant in a
532 multi-party bridge initiates a blind transfer, a Local channel will be used
533 to execute the dialplan location that the transferer sent the parties to. If
534 a participant in a multi-party bridge initiates an attended transfer,
535 several options are possible. If the attended transfer results in a transfer
536 to an application, a Local channel is used. If the attended transfer results
537 in a transfer to another channel, the resulting channels will be merged into
540 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
541 driver specific. If the channel variable is set on the transferrer channel,
542 the sound will be played to the target of an attended transfer.
544 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
545 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
546 listed. Any more peers in the bridge will not be included in the list.
547 BRIDGEPEER is not valid in holding bridges like parking since those channels
548 do not talk to each other even though they are in a bridge.
550 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
551 and will contain a value if the BRIDGEPEER's channel driver supports it.
553 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
554 was responsible for an attended transfer in a similar fashion to
557 * Modules using the Configuration Framework or Sorcery must have XML
558 configuration documentation. This configuration documentation is included
559 with the rest of Asterisk's XML documentation, and is accessible via CLI
560 commands. See the CLI changes for more information.
562 AMI (Asterisk Manager Interface)
564 * Major changes were made to both the syntax as well as the semantics of the
565 AMI protocol. In particular, AMI events have been substantially improved
566 in this version of Asterisk. For more information, please see the AMI
567 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
569 * AMI events that reference a particular channel or bridge will now always
570 contain a standard set of fields. When multiple channels or bridges are
571 referenced in an event, fields for at least some subset of the channels
572 and bridges in the event will be prefixed with a descriptive name to avoid
573 name collisions. See the AMI event documentation on the Asterisk wiki for
576 * The CLI command 'manager show commands' no longer truncates command names
577 longer than 15 characters and no longer shows authorization requirement
578 for commands. 'manager show command' now displays the privileges needed
579 for using a given manager command instead.
581 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
582 peer in its response if the peer has a subscribe context set.
584 * The SIPqualifypeer action now acknowledges the request once it has
585 established that the request is against a known peer. It also issues a new
586 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
588 * The PlayDTMF action now supports an optional 'Duration' parameter. This
589 specifies the duration of the digit to be played, in milliseconds.
591 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
592 updates when changes occur instead of requiring the use of pollmailboxes.
594 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
595 AMI client to manipulate audio currently being played back on a channel. The
596 supported operations depend on the application being used to send audio to
597 the channel. When the audio playback was initiated using the ControlPlayback
598 application or CONTROL STREAM FILE AGI command, the audio can be paused,
599 stopped, restarted, reversed, or skipped forward. When initiated by other
600 mechanisms (such as the Playback application), the audio can be stopped,
601 reversed, or skipped forward.
603 * Channel related events now contain a snapshot of channel state, adding new
604 fields to many of these events.
606 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
607 in a future release. Please use the common 'Exten' field instead.
609 * The AMI event 'UserEvent' from app_userevent now contains the channel state
610 fields. The channel state fields will come before the body fields.
612 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
613 'UnParkedCall' have changed significantly in the new res_parking module.
615 The 'Channel' and 'From' headers are gone. For the channel that was parked
616 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
617 has a number of fields associated with it. The old 'Channel' header relayed
618 the same data as the new 'ParkeeChannel' header.
620 The 'From' field was ambiguous and changed meaning depending on the event.
621 for most of these, it was the name of the channel that parked the call
622 (the 'Parker'). There is no longer a header that provides this channel name,
623 however the 'ParkerDialString' will contain a dialstring to redial the
624 device that parked the call.
626 On UnParkedCall events, the 'From' header would instead represent the
627 channel responsible for retrieving the parkee. It receives a channel
628 snapshot labeled 'Retriever'. The 'from' field is is replaced with
631 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
633 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
634 fashion has changed the field names 'StartExten' and 'StopExten' to
635 'StartSpace' and 'StopSpace' respectively.
637 * The deprecated use of | (pipe) as a separator in the channelvars setting in
638 manager.conf has been removed.
640 * Channel Variables conveyed with a channel no longer contain the name of the
641 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
642 ChanVariable: bar=baz. When multiple channels are present in a single AMI
643 event, the various ChanVariable fields will contain a suffix that specifies
644 which channel they correspond to.
646 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
647 event always conveys the AMI event for a particular channel.
649 * All 'Reload' events have been consolidated into a single event type. This
650 event will always contain a Module field specifying the name of the module
651 and a Status field denoting the result of the reload. All modules now issue
652 this event when being reloaded.
654 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
655 fail to receive this event due to being connected after modules have loaded.
656 AMI connections that want to know when Asterisk is ready should listen for
657 the 'FullyBooted' event.
659 * app_fax now sends the same send fax/receive fax events as res_fax. The
660 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
661 now the 'ReceiveFAX' event.
663 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
664 'MusicOnHoldStop'. The sub type field has been removed.
666 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
667 carrier for another protocol.
669 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
670 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
671 to the specific channel. 'Both' may be specified to play a tone to both
672 channels. The old 'yes' option is still accepted as a way of playing the
673 tone to Channel2 only.
675 * The AMI 'Status' response event to the AMI Status action replaces the
676 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
677 indicate what bridge the channel is currently in.
679 * The AMI 'Hold' event has been moved out of individual channel drivers, into
680 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
683 * The AMI events in app_queue have been made more consistent with each other.
684 Events that reference channels (QueueCaller* and Agent*) will show
685 information about each channel. The (infamous) 'Join' and 'Leave' AMI
686 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
688 * The 'MCID' AMI event now publishes a channel snapshot when available and
689 its non-channel-snapshot parameters now use either the "MCallerID" or
690 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
691 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
692 parameters in the channel snapshot.
694 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
695 'AgentLogin' and 'AgentLogoff' respectively.
697 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
698 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
700 * 'ChannelUpdate' events have been removed.
702 * All AMI events now contain a 'SystemName' field, if available.
704 * Local channel optimization is now conveyed in two events:
705 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
706 when the Local channel driver begins attempting to optimize itself out of
707 the media path; the End event is sent after the channel halves have
708 successfully optimized themselves out of the media path.
710 * Local channel information in events is now prefixed with 'LocalOne' and
711 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
712 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
713 and 'LocalOptimizationEnd' events.
715 * The option 'allowmultiplelogin' can now be set or overriden in a particular
716 account. When set in the general context, it will act as the default
717 setting for defined accounts.
719 * The 'BridgeAction' event was removed. It technically added no value, as the
720 Bridge Action already receives confirmation of the bridge through a
721 successful completion Event.
723 * The 'BridgeExec' events were removed. These events duplicated the events that
724 occur in the Briding API, and are conveyed now through BridgeCreate,
725 BridgeEnter, and BridgeLeave events.
727 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
728 previous versions. They now report all SR/RR packets sent/received, and
729 have been restructured to better reflect the data sent in a SR/RR. In
730 particular, the event structure now supports multiple report blocks.
732 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
733 raised when a blind transfer/attended transfer completes successfully.
734 They contain information about the transfer that just completed, including
735 the location of the transfered channel.
737 * Added a 'security' class to AMI which outputs the required fields for
738 security messages similar to the log messages from res_security_log
740 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
741 that describes the status value in a human readable string.
743 CDR (Call Detail Records)
745 * Significant changes have been made to the behavior of CDRs. The CDR engine
746 was effectively rewritten and built on the Stasis message bus. For a full
747 definition of CDR behavior in Asterisk 12, please read the specification
748 on the Asterisk wiki (wiki.asterisk.org).
750 * CDRs will now be created between all participants in a bridge. For each
751 pair of channels in a bridge, a CDR is created to represent the path of
752 communication between those two endpoints. This lets an end user choose who
753 to bill for what during bridge operations with multiple parties.
755 * The duration, billsec, start, answer, and end times now reflect the times
756 associated with the current CDR for the channel, as opposed to a cumulative
757 measurement of all CDRs for that channel.
759 * When a CDR is dispatched, user defined CDR variables from both parties are
760 included in the resulting CDR. If both parties have the same variable, only
761 the Party A value is provided.
763 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
764 information regarding the CDR engine is logged as verbose messages. This
765 option should only be used if the behavior of the CDR engine needs to be
768 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
769 normally configured in cdr.conf.
771 * Added CLI command 'cdr show active {channel}'. When {channel} is not
772 specified, this command provides a summary of the channels with CDR
773 information and their statistics. When {channel} is specified, it shows
774 detailed information about all records associated with {channel}.
776 CEL (Channel Event Logging)
778 * CEL has undergone significant rework in Asterisk 12, and is now built on the
779 Stasis message bus. Please see the specification for CEL on the Asterisk
780 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
783 * The 'extra' field of all CEL events that use it now consists of a JSON blob
784 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
786 * BLINDTRANSFER events now report the transferee bridge unique
787 identifier, extension, and context in a JSON blob as the extra string
788 instead of the transferee channel name as the peer.
790 * ATTENDEDTRANSFER events now report the peer as NULL and additional
791 information in the 'extra' string as a JSON blob. For transfers that occur
792 between two bridged channels, the 'extra' JSON blob contains the primary
793 bridge unique identifier, the secondary channel name, and the secondary
794 bridge unique identifier. For transfers that occur between a bridged channel
795 and a channel running an app, the 'extra' JSON blob contains the primary
796 bridge unique identifier, the secondary channel name, and the app name.
798 * LOCAL_OPTIMIZE events have been added to convey local channel
799 optimizations with the record occurring for the semi-one channel and
800 the semi-two channel name in the peer field.
802 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
803 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
804 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
805 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
806 regardless of whether or not that bridge happens to contain multiple
811 * When compiled with '--enable-dev-mode', the astobj2 library will now add
812 several CLI commands that allow for inspection of ao2 containers that
813 register themselves with astobj2. The CLI commands are 'astobj2 container
814 dump', 'astobj2 container stats', and 'astobj2 container check'.
816 * Added specific CLI commands for bridge inspection. This includes 'bridge
817 show all', which lists all bridges in the system, and 'bridge show {id}',
818 which provides specific information about a bridge.
820 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
821 ejecting the channels currently in the bridge. If the channels cannot
822 continue in the dialplan or application that put them in the bridge, they
825 * Added command 'bridge kick'. This will eject a single channel from a bridge.
827 * Added commands to inspect and manipulate the registered bridge technologies.
828 This include 'bridge technology show', which lists the registered bridge
829 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
830 which controls whether or not a registered bridge technology can be used
831 during smart bridge operations. If a technology is suspended, it will not
832 be used when a bridge technology is picked for channels; when unsuspended,
833 it can be used again.
835 * The command 'config show help {module} {type} {option}' will show
836 configuration documentation for modules with XML configuration
837 documentation. When {module}, {type}, and {option} are omitted, a listing
838 of all modules with registered documentation is displayed. When {module}
839 is specified, a listing of all configuration types for that module is
840 displayed, along with their synopsis. When {module} and {type} are
841 specified, a listing of all configuration options for that type are
842 displayed along with their synopsis. When {module}, {type}, and {option}
843 are specified, detailed information for that configuration option is
846 * Added 'core show sounds' and 'core show sound' CLI commands. These display
847 a listing of all installed media sounds available on the system and
848 detailed information about a sound, respectively.
850 * 'xmldoc dump' has been added. This CLI command will dump the XML
851 documentation DOM as a string to the specified file. The Asterisk core
852 will populate certain XML elements pulled from the source files with
853 additional run-time information; this command lets a user produce the
854 XML documentation with all information.
858 * Parking has been pulled from core and placed into a separate module called
859 res_parking. See Parking changes below for more details. Configuration for
860 parking should now be performed in res_parking.conf. Configuration for
861 parking in features.conf is now unsupported.
863 * Core attended transfers now have several new options. While performing an
864 attended transfer, the transferer now has the following options:
865 - *1 - cancel the attended transfer (configurable via atxferabort)
866 - *2 - complete the attended transfer, dropping out of the call
867 (configurable via atxfercomplete)
868 - *3 - complete the attended transfer, but stay in the call. This will turn
869 the call into a multi-party bridge (configurable via atxferthreeway)
870 - *4 - swap to the other party. Once an attended transfer has begun, this
871 options may be used multiple times (configurable via atxferswap)
873 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
874 must be on the channel initiating the transfer to have any effect.
876 * The BRIDGE_FEATURES channel variable would previously only set features for
877 the calling party and would set this feature regardless of whether the
878 feature was in caps or in lowercase. Use of a caps feature for a letter
879 will now apply the feature to the calling party while use of a lowercase
880 letter will apply that feature to the called party.
882 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
884 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
885 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
886 activated the dynamic feature.
888 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
889 only on the channel executing the dynamic feature. Executing a dynamic
890 feature on the bridge peer in a multi-party bridge will execute it on all
891 peers of the activating channel.
893 * You can now have the settings for a channel updated using the FEATURE()
894 and FEATUREMAP() functions inherited to child channels by setting
895 FEATURE(inherit)=yes.
897 * automixmon now supports additional channel variables from automon including:
898 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
899 and TOUCH_MIXMONITOR_MESSAGE_STOP
901 * A new general features.conf option 'recordingfailsound' has been added which
902 allowssetting a failure sound for a user tries to invoke a recording feature
903 such as automon or automixmon and it fails.
905 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
906 features.c for atxferdropcall=no to work properly. This option now just
911 * Added log rotation strategy 'none'. If set, no log rotation strategy will
912 be used. Given that this can cause the Asterisk log files to grow quickly,
913 this option should only be used if an external mechanism for log management
918 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
919 will store the path information for that peer when it registers. Realtime
920 tables can also use the 'supportpath' field to enable Path header support.
922 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
923 objectIdentifier. This maps to the supportpath option in sip.conf.
927 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
928 provides modules a useful abstraction on top of the many storage mechanisms
929 in Asterisk, including the Asterisk Database, static configuration files,
930 static Realtime, and dynamic Realtime. It also provides a caching service.
931 Users can configure a hierarchy of data storage layers for specific modules
934 * All future modules which utilize Sorcery for object persistence must have a
935 column named "id" within their schema when using the Sorcery realtime module.
936 This column must be able to contain a string of up to 128 characters in length.
938 Security Events Framework
940 * Security Event timestamps now use ISO 8601 formatted date/time instead of
941 the "seconds-microseconds" format that it was using previously.
945 * The Stasis message bus is a publish/subscribe message bus internal to
946 Asterisk. Many services in Asterisk are built on the Stasis message bus,
947 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
948 Stasis can be configured in stasis.conf. Note that these parameters operate
949 at a very low level in Asterisk, and generally will not require changes.
953 * When a channel driver is configured to enable jiterbuffers, they are now
954 applied unconditionally when a channel joins a bridge. If a jitterbuffer
955 is already set for that channel when it enters, such as by the JITTERBUFFER
956 function, then the existing jitterbuffer will be used and the one set by
957 the channel driver will not be applied.
961 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
962 dialplan applications provided by the app_agent_pool module. Agents are
963 connected with callers using the new AgentRequest dialplan application.
964 The Agents:<agent-id> device state is available to monitor the status of an
965 agent. See agents.conf.sample for valid configuration options.
967 * The updatecdr option has been removed. Altering the names of channels on a
968 CDR is not supported - the name of the channel is the name of the channel,
969 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
970 has also been removed, for the same reason.
972 * The endcall and enddtmf configuration options are removed. Use the
973 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
974 channel before calling AgentLogin.
978 * chan_bridge has been removed. Its functionality has been incorporated
979 directly into the ConfBridge application itself.
983 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
984 of the specified span and its B-channels. Note that this command should
985 only be used if you understand the risks it entails.
987 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
988 A range of channels can be specified to be destroyed. Note that this command
989 should only be used if you understand the risks it entails.
991 * Added the CLI command 'dahdi create channels'. A range of channels can be
992 specified to be created, or the keyword 'new' can be used to add channels
995 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
996 the exact configured mailbox name. For app_voicemail mailboxes this is
999 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1003 * IPv6 support has been added. We are now able to bind to and
1004 communicate using IPv6 addresses.
1008 * The /b option has been removed.
1010 * chan_local moved into the system core and is no longer a loadable module.
1014 * Added general support for busy detection.
1016 * Added ECAM command support for Sony Ericsson phones.
1020 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1021 SIP stack. A collection of resource modules provides the bulk of the SIP
1022 functionality. For more information on the new SIP channel driver, see
1023 https://wiki.asterisk.org/wiki/x/JYGLAQ
1027 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1028 using the 'supportpath' setting, either on a global basis or on a peer basis.
1029 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1030 set of proxies by using a pre-loaded route-set defined by the Path headers in
1031 the REGISTER request. See Realtime updates for more configuration information.
1033 * The SIP_CODEC family of variables may now specify more than one codec. Each
1034 codec must be separated by a comma. The first codec specified is the
1035 preferred codec for the offer. This allows a dialplan writer to specify both
1036 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1038 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1039 in the core, and can be filtered out using the 'eventfilter' parameter
1042 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1043 codecs configured for a peer instead of the requested codec.
1045 * The option "register_retry_403" has been added to chan_sip to work around
1046 servers that are known to erroneously send 403 in response to valid
1047 REGISTER requests and allows Asterisk to continue attepmting to connect.
1051 * Added the 'immeddialkey' parameter. If set, when the user presses the
1052 configured key the already entered number will be immediately dialed. This
1053 is useful when the dialplan allows for variable length pattern matching.
1054 Valid options are '*' and '#'.
1056 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1057 milliseconds) before a call forward is considered to not be answered.
1059 * The 'serviceurl' parameter allows Service URLs to be attached to line
1068 * The password option has been disabled, as the AgentLogin application no
1069 longer provides authentication.
1073 * Due to changes in the Asterisk core, this function is no longer needed to
1074 preserve a MixMonitor on a channel during transfer operations and dialplan
1075 execution. It is effectively obsolete.
1079 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1080 deprecated. Use the CHANNEL function instead to access these attributes.
1082 * The 'l' option has been removed. When reading a CDR attribute, the most
1083 recent record is always used. When writing a CDR attribute, all non-finalized
1086 * The 'r' option has been removed, for the same reason as the 'l' option.
1088 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1093 * A new function CDR_PROP has been added. This function lets you set properties
1094 on a channel's active CDRs. This function is write-only. Properties accept
1095 boolean values to set/clear them on the channel's CDRs. Valid properties
1097 - 'party_a' - make this channel the preferred Party A in any CDR between two
1098 channels. If two channels have this property set, the creation time of the
1099 channel is used to determine who is Party A. Note that dialed channels are
1100 never Party A in a CDR.
1101 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1102 application when set to True, and analogous to the 'e' option in ResetCDR
1107 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1108 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1109 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1112 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1113 string, i.e., [[context],extension],priority. If set on a channel, if a
1114 channel leaves a bridge but is not hung up it will resume dialplan execution
1119 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1120 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1121 The value of this setting is ignored when disabled is used for the argument.
1125 * A new function provided by chan_pjsip, this function can be used in
1126 conjunction with the Dial application to construct a dial string that will
1127 dial all contacts on an Address of Record associated with a chan_pjsip
1132 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1133 outbound channel prior to dialing.
1137 * Redirecting reasons can now be set to arbitrary strings. This means
1138 that the REDIRECTING dialplan function can be used to set the redirecting
1139 reason to any string. It also allows for custom strings to be read as the
1140 redirecting reason from SIP Diversion headers.
1144 * The SPEECH_ENGINE function now supports read operations. When read from, it
1145 will return the current value of the requested attribute.
1149 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1150 system as mailbox@context. The rest of the system cannot add @default
1151 to mailbox identifiers for app_voicemail that do not specify a context
1152 any longer. It is a mailbox identifier format that should only be
1153 interpreted by app_voicemail.
1159 res_agi (Asterisk Gateway Interface)
1161 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1163 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1166 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1167 will start the playback of the audio at the position specified. It will
1168 also return the final position of the file in 'endpos'.
1170 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1171 channel variable if the user stopped the file playback or if a remote
1172 entity stopped the playback. If neither stopped the playback, it will
1173 indicate the overall success/failure of the playback. If stopped early,
1174 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1177 * The SAY ALPHA command now accepts an additional parameter to control
1178 whether it specifies the case of uppercase, lowercase, or all letters to
1179 provide functionality similar to SayAlphaCase.
1181 res_ari (Asterisk RESTful Interface) (and others)
1183 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1184 control telephony primitives in Asterisk by remote client. This includes
1185 channels, bridges, endpoints, media, and other fundamental concepts. Users
1186 of ARI can develop their own communications applications, controlling
1187 multiple channels using an HTTP RESTful interface and receiving JSON events
1188 about the objects via a WebSocket connection. ARI can be configured in
1189 Asterisk via ari.conf. For more information on ARI, see
1190 https://wiki.asterisk.org/wiki/x/0YCLAQ
1194 * Parking has been extracted from the Asterisk core as a loadable module,
1195 res_parking. Configuration for parking is now provided by res_parking.conf.
1196 Configuration through features.conf is no longer supported.
1198 * res_parking uses the configuration framework. If an invalid configuration is
1199 supplied, res_parking will fail to load or fail to reload. Previously,
1200 invalid configurations would generally be accepted, with certain errors
1201 resulting in individually disabled parking lots.
1203 * Parked calls are now placed in bridges. While this is largely an
1204 architectural change, it does have implications on how channels in a parking
1205 lot are viewed. For example, commands that display channels in bridges will
1206 now also display the channels in a parking lot.
1208 * The order of arguments for the new parking applications have been modified.
1209 Timeout and return context/exten/priority are now implemented as options,
1210 while the name of the parking lot is now the first parameter. See the
1211 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1212 in-depth information as well as syntax.
1214 * Extensions are by default no longer automatically created in the dialplan to
1215 park calls or pickup parked calls. Generation of dialplan extensions can be
1216 enabled using the 'parkext' configuration option.
1218 * ADSI functionality for parking is no longer supported. The 'adsipark'
1219 configuration option has been removed as a result.
1221 * The PARKINGSLOT channel variable has been deprecated in favor of
1222 PARKING_SPACE to match the naming scheme of the new system.
1224 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1225 channel even when the configuration option 'comebactoorigin' is enabled.
1227 * A new CLI command 'parking show' has been added. This allows a user to
1228 inspect the parking lots that are currently in use.
1229 'parking show <parkinglot>' will also show the parked calls in a specific
1232 * The CLI command 'parkedcalls' is now deprecated in favor of
1233 'parking show <parkinglot>'.
1235 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1236 can be used to get a list of parked calls for a specific parking lot.
1238 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1239 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1240 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1241 longer a required argument.
1243 * The ParkAndAnnounce application is now provided through res_parking instead
1244 of through the separate app_parkandannounce module.
1246 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1247 by default. Instead, it will follow the timeout rules of the parking lot. The
1248 old behavior can be reproduced by using the 'c' option.
1250 * Dynamic parking lots will now fail to be created under the following
1252 - if the parking lot specified by PARKINGDYNAMIC does not exist
1253 - if they require exclusive park and parkedcall extensions which overlap
1254 with existing parking lots.
1256 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1257 currently contain no calls. Dynamic parking lots containing parked calls
1258 will persist through the reloads without alteration.
1260 * If 'parkext_exclusive' is set for a parking lot and that extension is
1261 already in use when that parking lot tries to register it, this is now
1262 considered a parking system configuration error. Configurations which do
1263 this will be rejected.
1265 * Added channel variable PARKER_FLAT. This contains the name of the extension
1266 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1267 comebacktoorigin is disabled, but the dialplan or an external control
1268 mechanism wants to use the extension in the park-dial context that was
1269 generated to re-dial the parker on timeout.
1271 res_pjsip (and many others)
1273 * A large number of resource modules make up the SIP stack based on pjsip.
1274 The chan_pjsip channel driver users these resource modules to provide
1275 various SIP functionality in Asterisk. The majority of configuration for
1276 these modules is performed in pjsip.conf. Other modules may use their
1277 own configuration files.
1279 * Added 'set_var' option for an endpoint. For each variable specified that
1280 variable gets set upon creation of a channel involving the endpoint.
1284 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1285 them, an Asterisk-specific version of PJSIP needs to be installed.
1286 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1288 res_statsd/res_chan_stats
1290 * A new resource module, res_statsd, has been added, which acts as a statsd
1291 client. This module allows Asterisk to publish statistics to a statsd
1292 server. In conjunction with res_chan_stats, it will publish statistics about
1293 channels to the statsd server. It can be configured via res_statsd.conf.
1297 * Device state for XMPP buddies is now available using the following format:
1298 XMPP/<client name>/<buddy address>
1299 If any resource is available the device state is considered to be not in use.
1300 If no resources exist or all are unavailable the device state is considered
1307 Realtime/Database Scripts
1309 * Asterisk previously included example db schemas in the contrib/realtime/
1310 directory of the source tree. This has been replaced by a set of database
1311 migrations using the Alembic framework. This allows you to use alembic to
1312 initialize the database for you. It will also serve as a database migration
1313 tool when upgrading Asterisk in the future.
1315 See contrib/ast-db-manage/README.md for more details.
1319 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1320 This python script will convert an existing sip.conf file to a
1321 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1322 is meant to be an aid in converting an existing chan_sip configuration to
1323 a chan_pjsip configuration, but it is expected that configuration beyond
1324 what the script provides will be needed.
1327 ------------------------------------------------------------------------------
1328 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1329 ------------------------------------------------------------------------------
1333 * The Asterisk build system will now build and install a shared library
1334 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1335 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1336 that Asterisk can ensure that these functions do *not* get called by any
1337 modules that are loaded into Asterisk, since they should only be called once
1338 in any single process. If desired, this feature can be disabled by supplying
1339 the "--disable-asteriskssl" option to the configure script.
1341 * A new make target, 'full', has been added to the Makefile. This performs
1342 the same compilation actions as make all, but will also scan the entirety of
1343 each source file for documentation. This option is needed to generate AMI
1344 event documentation. Note that your system must have Python in order for
1345 this make target to succeed.
1347 * The optimization portion of the build system has been reworked to avoid
1348 broken builds on certain architectures. All architecture-specific
1349 optimization has been removed in favor of using -march=native to allow gcc
1350 to detect the environment in which it is running when possible. This can
1351 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1353 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1354 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1356 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1357 previously parsed the header file to obtain the version of Asterisk, you
1358 will now have to go through Asterisk to get the version information.
1366 * Added 'F()' option. Similar to the dial option, this can be supplied with
1367 arguments indicating where the callee should go after the caller is hung up,
1368 or without options specified, the priority after the Queue will be used.
1373 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1374 all non-admin participants on a conference. The confbridge configuration
1375 file also allows for the default sounds played to all conference users when
1376 this occurs to be overriden using sound_participants_unmuted and
1377 sound_participants_muted.
1379 * Added menu action participant_count. This will playback the number of
1380 current participants in a conference.
1382 * Added announcement configuration option to user profile. If set the sound
1383 file will be played to the user, and only the user, upon joining the
1386 * Added record_file_append option that defaults to "yes", but if set to no
1387 will create a new file between each start/stop recording.
1392 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1393 channels respectively before the callee channels are called.
1398 * Added support for IPv6.
1400 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1401 external process will cause the current playlist to be cleared, including
1402 stopping any audio file that is currently playing. This is useful when you
1403 want to interrupt audio playback only when specific DTMF is entered by the
1409 * A new option, 'I' has been added to app_followme. By setting this option,
1410 Asterisk will not update the caller with connected line changes when they
1411 occur. This is similar to app_dial and app_queue.
1413 * The 'N' option is now ignored if the call is already answered.
1415 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1416 and caller channels respectively before the callee channels are called.
1418 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1424 * MixMonitor hooks now have IDs associated with them which can be used to
1425 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1426 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1427 now accepts that ID as an argument.
1429 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1430 indicated mailboxes.
1435 * The connect action in app_mysql now allows you to specify a port number to
1436 connect to. This is useful if you run a MySQL server on a non-standard
1442 * Increased the default number of allowed destinations from 5 to 12.
1447 * The app_page application now no longer depends on DAHDI or app_meetme. It
1448 has been re-architected to use app_confbridge internally.
1453 * Added queue options autopausebusy and autopauseunavail for automatically
1454 pausing a queue member when their device reports busy or congestion.
1456 * The 'ignorebusy' option for queue members has been deprecated in favor of
1457 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1458 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1459 per interface basis. Individual ringinuse values can now be set in
1460 queues.conf via an argument to member definitions. Lastly, the queue
1461 'ringinuse' setting now only determines defaults for the per member
1462 'ringinuse' setting and does not override per member settings like it does
1463 in earlier versions.
1465 * Added 'F()' option. Similar to the dial option, this can be supplied with
1466 arguments indicating where the callee should go after the caller is hung up,
1467 or without options specified, the priority after the Queue will be used.
1469 * Added new option log_member_name_as_agent, which will cause the membername to
1470 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1471 state_interface has been set.
1473 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1475 * App_queue will now play periodic announcements for the caller that
1476 holds the first position in the queue while waiting for answer.
1480 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1481 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1482 changed arguments to SayUnixTime so that every option is truly optional even
1483 when using multiple options (so that j option could be used without having to
1484 manually specify timezone and format) There are other benefits, e.g., format
1485 can now be used without specifying time zone as well.
1490 * Addition of the VM_INFO function - see Function changes.
1492 * The imapserver, imapport, and imapflags configuration options can now be
1493 overriden on a user by user basis.
1495 * When voicemail plays a message's envelope with saycid set to yes, when
1496 reaching the caller id field it will play a recording of a file with the same
1497 base name as the sender's callerid if there is a similarly named file in
1498 <astspooldir>/recordings/callerids/
1500 * Voicemails now contains a unique message identifier "msg_id", which is stored
1501 in the message envelope with the sound files. IMAP backends will now store
1502 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1503 backends will store the message identifier in a "msg_id" column. See
1504 UPGRADE.txt for more information.
1506 * Added VoiceMailPlayMsg application. This application will play a single
1507 voicemail message from a mailbox. The result of the application, SUCCESS or
1508 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1513 * Hangup handlers can be attached to channels using the CHANNEL() function.
1514 Hangup handlers will run when the channel is hung up similar to the h
1515 extension. The hangup_handler_push option will push a GoSub compatible
1516 location in the dialplan onto the channel's hangup handler stack. The
1517 hangup_handler_pop option will remove the last added location, and optionally
1518 replace it with a new GoSub compatible location. The hangup_handler_wipe
1519 option will remove all locations on the stack, and optionally add a new
1522 * The expression parser now recognizes the ABS() absolute value function,
1523 which will convert negative floating point values to positive values.
1525 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1526 control of faxdetect.
1528 * Addition of the VM_INFO function that can be used to retrieve voicemail
1529 user information, such as the email address and full name.
1530 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1533 * The REDIRECTING function now supports the redirecting original party id
1536 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1537 lets you set some of the configuration options from the [general] section
1538 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1539 the key sequence used to activate built-in features, such as blindxfer,
1540 and automon. See the built-in documentation for details.
1542 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1543 instead of simply the uri. This is the format that MessageSend() can use
1544 in the from parameter for outgoing SIP messages.
1546 * Added the PRESENCE_STATE function. This allows retrieving presence state
1547 information from any presence state provider. It also allows setting
1548 presence state information from a CustomPresence presence state provider.
1549 See AMI/CLI changes for related commands.
1551 * Added the AMI_CLIENT function to make manager account attributes available
1552 to the dialplan. It currently supports returning the current number of
1553 active sessions for a given account.
1555 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1556 and the REDIRECTING functions.
1564 * Added a manager event "LocalBridge" for local channel call bridges between
1565 the two pseudo-channels created.
1570 * Added dialtone_detect option for analog ports to disconnect incoming
1571 calls when dialtone is detected.
1573 * Added option colp_send to send ISDN connected line information. Allowed
1574 settings are block, to not send any connected line information; connect, to
1575 send connected line information on initial connect; and update, to send
1576 information on any update during a call. Default is update.
1578 * Add options namedcallgroup and namedpickupgroup to support installations
1579 where a higher number of groups (>64) is required.
1581 * Added support to use private party ID information with PRI calls.
1586 * A new channel driver named chan_motif has been added which provides support for
1587 Google Talk and Jingle in a single channel driver. This new channel driver includes
1588 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1589 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1590 specification, current Google Jingle specification, and the original Google Talk
1596 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1597 globally and overriden on a peer by peer basis.
1599 * Direct media functionality has been added. Options in config are:
1600 directmedia (directrtp) and directrtpsetup (earlydirect)
1602 * ChannelUpdate events now contain a CallRef header.
1607 * Asterisk will no longer substitute CID number for CID name in the display
1608 name field if CID number exists without a CID name. This change improves
1609 compatibility with certain device features such as Avaya IP500's directory
1612 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1613 created using that setting to not be removed during SIP reload.
1615 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1616 request with a "Record:" header, this will turn the requested feature on/off.
1617 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1618 dynamic features must be enabled and configured properly on the requesting
1619 channel for this to function properly.
1621 * Add support to realtime for the 'callbackextension' option.
1623 * When multiple peers exist with the same address, but differing
1624 callbackextension options, incoming requests that are matched by address
1625 will be matched to the peer with the matching callbackextension if it is
1628 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1629 which set the force_rport and comedia options automatically if Asterisk
1630 detects that an incoming SIP request crossed a NAT after being sent by
1631 the remote endpoint.
1633 * The default global nat setting in sip.conf has been changed from force_rport
1634 to auto_force_rport.
1636 * NAT settings are now a combinable list of options. The equivalent of the
1637 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1639 * Adds an option send_diversion which can be disabled to prevent
1640 diversion headers from automatically being added to INVITE requests.
1642 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1643 be sent to the remote host at a given interval to keep the NAT mapping open.
1644 This can be enabled using the keepalive configuration option.
1646 * Add option 'tonezone' to specify country code for indications. This option
1647 can be set both globally and overridden for specific peers.
1649 * The SIP Security Events Framework now supports IPv6.
1651 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1652 between multiple user agents. When set, for directmedia reinvites,
1653 Asterisk will not send an immediate reinvite on an incoming call leg. This
1654 option is useful when peered with another SIP user agent that is known to
1655 send immediate direct media reinvites upon call establishment.
1657 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1660 * Add options subminexpiry and submaxexpiry to set limits of subscription
1661 timer independently from registration timer settings. The setting of the
1662 registration timer limits still is done by options minexpiry, maxexpiry
1663 and defaultexpiry. For backwards compatibility the setting of minexpiry
1664 and maxexpiry also is used to configure the subscription timer limits if
1665 subminexpiry and submaxexpiry are not set in sip.conf.
1667 * Set registration timer limits to default values when reloading sip
1668 configuration and values are not set by configuration.
1670 * Add options namedcallgroup and namedpickupgroup to support installations
1671 where a higher number of groups (>64) is required.
1673 * When a MESSAGE request is received, the address the request was received from
1674 is now saved in the SIP_RECVADDR variable.
1676 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1677 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1678 the ANI2/OLI information is set on the channel, which can be retrieved using
1679 the CALLERID function.
1681 * Peers can now be configured to support negotiation of ICE candidates using
1682 the setting icesupport. See res_rtp_asterisk changes for more information.
1684 * Added support for format attribute negotiation. See the Codecs changes for
1687 * Extra headers specified with SIPAddHeader are sent with the REFER message
1688 when using Transfer application. See refer_addheaders in sip.conf.sample.
1690 * Added support to use private party ID information with calls.
1692 * Adds an option discard_remote_hold_retrieval that when set stops telling
1693 the peer to start music on hold.
1698 * Added skinny version 17 protocol support.
1702 --------------------
1703 * Added ability to use multiple lines for a single phone. This allows multiple
1704 calls to occur on a single phone, using callwaiting and switching between calls.
1706 * Added option 'sharpdial' allowing end dialing by pressing # key
1708 * Added option 'interdigit_timer' to control phone dial timeout
1710 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1712 * Added global 'debug' option, that enables debug in channel driver
1714 * Added ability to translate on-screen menu in multiple languages. Tested on
1715 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1716 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1719 * In addition to English added French and Russian languages for on-screen menus
1721 * Reworked dialing number input: added dialing by timeout, immediate dial on
1722 on dialplan compare, phone number length now not limited by screen size
1724 * Added ability to pickup a call using features.conf defined value and
1730 * Add options namedcallgroup and namedpickupgroup to support installations
1731 where a higher number of groups (>64) is required.
1733 * Added support to use private party ID information with calls.
1738 * The minimum DTMF duration can now be configured in asterisk.conf
1739 as "mindtmfduration". The default value is (as before) set to 80 ms.
1740 (previously it was only available in source code)
1742 * Named ACLs can now be specified in acl.conf and used in configurations that
1743 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1744 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1745 working ACL. In addition, some CLI commands have been added to provide
1746 show information and allow for module reloading - see CLI Changes.
1748 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1749 items (separated by commas), and items in the rule can be negated by prefixing
1750 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1751 longer necessray to control the order that the 'permit' and 'deny' columns are
1752 returned from queries.
1754 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1755 be used within the dynamic weight attribute when specifying a mapping.
1757 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1758 header, instead of putting the user defined event name there. When enabled
1759 the UserDefType header is added for user defined events. This feature is
1760 enabled with the setting show_user_defined.
1762 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1763 line purposes use the following variables instead of their macro equivalents:
1764 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1765 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1766 cc_callback_macro in channel configurations.
1768 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1771 * Call files now support the "early_media" option to connect with an outgoing
1772 extension when early media is received.
1774 * Added support to use private party ID information with calls.
1779 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1780 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1781 AGI application would exit immediately after a channel hangup is detected.
1783 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1784 are resolved and each address is attempted in turn until one succeeds or
1788 AMI (Asterisk Manager Interface)
1790 * The originate action now has an option "EarlyMedia" that enables the
1791 call to bridge when we get early media in the call. Previously,
1792 early media was disregarded always when originating calls using AMI.
1794 * Added setvar= option to manager accounts (much like sip.conf)
1796 * Originate now generates an error response if the extension given is not found
1799 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1800 them if the i(variable) option is used. StopMixMonitor will accept
1801 MixMonitorID as an option to close specific MixMonitors.
1803 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1804 updated to include information about peers configured with
1805 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1806 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1807 returned if auto_force_rport is not enabled.
1809 * Added SIPpeerstatus manager command which will generate PeerStatus events
1810 similar to the existing PeerStatus events found in chan_sip on demand.
1812 * Hangup now can take a regular expression as the Channel option. If you want
1813 to hangup multiple channels, use /regex/ as the Channel option. Existing
1814 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1815 the manager will send you a list of channels back that were hung up.
1817 * Support for IPv6 addresses has been added.
1819 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1820 documentation is only generated when Asterisk is compiled using 'make full'.
1821 See the CLI section for commands to display AMI event information.
1823 * The AMI Hangup event now includes the AccountCode header so you can easily
1824 correlate with AMI Newchannel events.
1826 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1827 the StateInterface of the queue member.
1829 * Added AMI event SessionTimeout in the Call category that is issued when a
1830 call is terminated due to either RTP stream inactivity or SIP session timer
1833 * CEL events can now contain a user defined header UserDefType. See core
1834 changes for more information.
1836 * OOH323 ChannelUpdate events now contain a CallRef header.
1838 * Added PresenceState command. This command will report the presence state for
1839 the given presence provider.
1841 * Added Parkinglots command. This will list all parking lots as a series of
1842 AMI Parkinglot events.
1844 * Added MessageSend command. This behaves in the same manner as the
1845 MessageSend application, and is a technolgoy agnostic mechanism to send out
1846 of call text messages.
1848 * Added "message" class authorization. This grants an account permission to
1849 send out of call messages. Write-only.
1854 * The "dialplan add include" command has been modified to create context a context
1855 if one does not already exist. For instance, "dialplan add include foo into bar"
1856 will create context "bar" if it does not already exist.
1858 * A "dialplan remove context" command has been added to remove a context from
1861 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1862 filenames of all running mixmonitors on a channel.
1864 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1865 numeric instead of 0, 1, or 2.
1867 * "stun show status" will show a table describing how the STUN client is
1870 * "acl show [named acl]" will show information regarding a Named ACL. The
1871 acl module can be reloaded with "reload acl".
1873 * Added CLI command to display AMI event information - "manager show events",
1874 which shows a list of all known and documented AMI events, and "manager show
1875 event [event name]", which shows detail information about a specific AMI
1878 * The result of the CLI command "queue show" now includes the state interface
1879 information of the queue member.
1881 * The command "core set verbose" will now set a separate level of logging for
1882 each remote console without affecting any other console.
1884 * Added command "cdr show pgsql status" to check connection status
1886 * "sip show channel" will now display the complete route set.
1888 * Added "presencestate list" command. This command will list all custom
1889 presence states that have been set by using the PRESENCE_STATE dialplan
1892 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1893 command. This changes a custom presence to a new state.
1898 * Codec lists may now be modified by the '!' character, to allow succinct
1899 specification of a list of codecs allowed and disallowed, without the
1900 requirement to use two different keywords. For example, to specify all
1901 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1903 * Add support for parsing SDP attributes, generating SDP attributes, and
1904 passing it through. This support includes codecs such as H.263, H.264, SILK,
1905 and CELT. You are able to set up a call and have attribute information pass.
1906 This should help considerably with video calls.
1908 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1909 just like the GSM codec.
1913 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1914 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1918 * Asterisk version and build information is now logged at the beginning of a
1921 * Threads belonging to a particular call are now linked with callids which get
1922 added to any log messages produced by those threads. Log messages can now be
1923 easily identified as involved with a certain call by looking at their call id.
1924 Call ids may also be attached to log messages for just about any case where
1925 it can be determined to be related to a particular call.
1927 * Each logging destination and console now have an independent notion of the
1928 current verbosity level. Logger.conf now allows an optional argument to
1929 the 'verbose' specifier, indicating the level of verbosity sent to that
1930 particular logging destination. Additionally, remote consoles now each
1931 have their own verbosity level. The command 'core set verbose' will now set
1932 a separate level for each remote console without affecting any other
1938 * Added 'announcement' option which will play at the start of MOH and between
1939 songs in modes of MOH that can detect transitions between songs (eg.
1945 * New per parking lot options: comebackcontext and comebackdialtime. See
1946 configs/features.conf.sample for more details.
1948 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1951 * Channel variable PARKEDCALL is now set with the name of the parking lot
1952 when a timeout occurs.
1958 CDR Postgresql Driver
1960 * Added command "cdr show pgsql status" to check connection status
1963 CDR Adaptive ODBC Driver
1965 * Added schema option for databases that support specifying a schema.
1973 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1974 CALENDAR_WRITE has completed successfully.
1979 * A new option, 'probation' has been added to rtp.conf
1980 RTP in strictrtp mode can now require more than 1 packet to exit learning
1981 mode with a new source (and by default requires 4). The probation option
1982 allows the user to change the required number of packets in sequence to any
1983 desired value. Use a value of 1 to essentially restore the old behavior.
1984 Also, with strictrtp on, Asterisk will now drop all packets until learning
1985 mode has successfully exited. These changes are based on how pjmedia handles
1986 media sources and source changes.
1988 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1989 enabled or disabled using the icesupport setting. A variety of other
1990 settings have been introduced to configure STUN/TURN connections.
1995 * A new module, res_corosync, has been introduced. This module uses the
1996 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1997 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1998 Device State (presence) information. This module is very similar to, and
1999 is a replacement for the res_ais module that was in previous releases of
2005 * This module adds a cleaned up, drop-in replacement for res_jabber called
2006 res_xmpp. This provides the same externally facing functionality but is
2007 implemented differently internally. res_jabber has been deprecated in favor
2008 of res_xmpp; please see the UPGRADE.txt file for more information.
2013 * The safe_asterisk script has been updated to allow several of its parameters
2014 to be set from environment variables. This also enables a custom run
2015 directory of Asterisk to be specified, instead of defaulting to /tmp.
2017 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2018 its value to determine the directory to assume is the top-level directory of
2019 the source tree. If the variable is not set, it defaults to the current
2020 behavior and uses the current working directory.
2022 ------------------------------------------------------------------------------
2023 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2024 ------------------------------------------------------------------------------
2028 * Asterisk now has protocol independent support for processing text messages
2029 outside of a call. Messages are routed through the Asterisk dialplan.
2030 SIP MESSAGE and XMPP are currently supported. There are options in
2031 jabber.conf and sip.conf to allow enabling these features.
2032 -> jabber.conf: see the "sendtodialplan" and "context" options.
2033 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2034 and "outofcall_message_context" options.
2035 The MESSAGE() dialplan function and MessageSend() application have been
2036 added to go along with this functionality. More detailed usage information
2037 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2038 * If real-time text support (T.140) is negotiated, it will be preferred for
2039 sending text via the SendText application. For example, via SIP, messages
2040 that were once sent via the SIP MESSAGE request would be sent via RTP if
2041 T.140 text is negotiated for a call.
2045 * parkedmusicclass can now be set for non-default parking lots.
2047 Asterisk Manager Interface
2048 --------------------------
2049 * PeerStatus now includes Address and Port.
2050 * Added Hold events for when the remote party puts the call on and off hold
2051 for chan_dahdi ISDN channels.
2052 * Added new action MeetmeListRooms to list active conferences (shows same
2053 data as "meetme list" at the CLI).
2054 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2055 Description field that is set by 'description' in the channel configuration
2057 * Added Uniqueid header to UserEvent.
2058 * Added new action FilterAdd to control event filters for the current session.
2059 This requires the system permission and uses the same filter syntax as
2060 filters that can be defined in manager.conf
2061 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2062 versions had some instances of the event converted, but others were left
2063 as-is. All Unlink events should now be converted to Bridge events. The AMI
2064 protocol version number was incremented to 1.2 as a result of this change.
2066 Asterisk HTTP Server
2067 --------------------------
2068 * The HTTP Server can bind to IPv6 addresses.
2071 --------------------------
2072 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2073 with busydetect. usage example: busypattern=200,200,200,600
2076 --------------------------
2077 * New 'gtalk show settings' command showing the current settings loaded from
2079 * The 'logger reload' command now supports an optional argument, specifying an
2080 alternate configuration file to use.
2081 * 'dialplan add extension' command will now automatically create a context if
2082 the specified context does not exist with a message indicated it did so.
2083 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2084 Description field which can be populated with 'description' in the channel
2085 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2088 --------------------------
2089 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2090 thus allowing records which do NOT match the specified filter.
2091 * Added ability to log CONGESTION calls to CDR
2094 --------------------------
2095 * Ability to define custom SILK formats in codecs.conf.
2096 * Addition of speex32 audio format with translation.
2097 * CELT codec pass-through support and ability to define
2098 custom CELT formats in codecs.conf.
2099 * Ability to read raw signed linear files with sample rates
2100 ranging from 8khz - 192khz. The new file extensions introduced
2101 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2102 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2103 Skinny, H.323, etc) can still only support the following codecs:
2104 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2105 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2106 Video: h261, h263, h263p, h264, mpeg4
2111 --------------------------
2112 * New highly optimized and customizable ConfBridge application capable of
2113 mixing audio at sample rates ranging from 8khz-96khz.
2114 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2115 and bridge profiles on a channel.
2116 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2117 about a conference such as locked status and number of parties, admins,
2119 * Addition of video_mode option in confbridge.conf for adding video support
2120 into a bridge profile.
2121 * Addition of the follow_talker video_mode in confbridge.conf. This video
2122 mode dynamically switches the video feed to always display the loudest talker
2123 supplying video in the conference.
2127 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2128 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2129 variables from asterisk.conf.
2133 * Addition of the JITTERBUFFER dialplan function. This function allows
2134 for jitterbuffering to occur on the read side of a channel. By using
2135 this function conference applications such as ConfBridge and MeetMe can
2136 have the rx streams jitterbuffered before conference mixing occurs.
2137 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2139 * Added STRREPLACE function. This function let's the user search a variable
2140 for a given string to replace with another string as many times as the
2141 user specifies or just throughout the whole string.
2142 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2143 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2144 * Added extensions to chan_ooh323 in function CHANNEL()
2146 libpri channel driver (chan_dahdi) DAHDI changes
2147 --------------------------
2148 * Added moh_signaling option to specify what to do when the channel's bridged
2149 peer puts the ISDN channel on hold.
2150 * Added display_send and display_receive options to control how the display ie
2151 is handled. To send display text from the dialplan use the SendText()
2152 application when the option is enabled.
2153 * Added mcid_send option to allow sending a MCID request on a span.
2156 --------------------------
2157 * Added setvar option to calendar.conf to allow setting channel variables on
2158 notification channels.
2159 * Added "calendar show types" CLI command to list registered calendar
2163 --------------------------
2164 * Added two new options, r and t with file name arguments to record
2165 single direction (unmixed) audio recording separate from the bidirectional
2166 (mixed) recording. The mixed file name argument is optional now as long
2167 as at least one recording option is used.
2170 --------------------------
2171 * Added a new option, l, which will disable local call optimization for
2172 channels involved with the FollowMe thread. Use this option to improve
2173 compatability for a FollowMe call with certain dialplan apps, options, and
2177 --------------------------
2178 * Added option "k" that will automatically close the conference when there's
2179 only one person left when a user exits the conference.
2182 --------------------------
2183 * cel_pgsql now supports the 'extra' column for data added using the
2184 CELGenUserEvent() application.
2187 --------------------------
2188 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2189 in the sample extensions.lua file for syntax details.
2190 * Applications that perform jumps in the dialplan such as Goto will now
2191 execute properly. When pbx_lua detects that the context, extension, or
2192 priority we are executing on has changed it will immediately return control
2193 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2194 the priority after the currently executing priority.
2195 * An autoservice is now started by default for pbx_lua channels. It can be
2196 stopped and restarted using the autoservice_stop() and autoservice_start()
2200 --------------------------
2201 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2202 into a FAXStatus event with an 'Operation' header that will be either
2203 'send', 'receive', and 'gateway'.
2204 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2205 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2206 feature will handle converting a fax call between an audio T.30 fax terminal
2207 and an IFP T.38 fax terminal.
2211 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2212 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2213 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2217 * Added general option negative_penalty_invalid default off. when set
2218 members are seen as invalid/logged out when there penalty is negative.
2219 for realtime members when set remove from queue will set penalty to -1.
2220 * Added queue option autopausedelay when autopause is enabled it will be
2221 delayed for this number of seconds since last successful call if there
2222 was no prior call the agent will be autopaused immediately.
2223 * Added member option ignorebusy this when set and ringinuse is not
2224 will allow per member control of multiple calls as ringinuse does for
2229 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2231 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2232 one participant left (much like a normal call bridge)
2233 * Added extra argument to Originate to set timeout.
2237 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2238 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2239 utility in the UTILS section of menuselect. If an existing astdb is found and no
2240 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2241 convert an existing astdb to the SQLite3 version automatically at runtime.
2245 * Modules marked as deprecated are no longer marked as building by default. Enabling
2246 these modules is still available via menuselect.
2250 * authdebug is now disabled by default. To enable this functionaility again
2251 set authdebug = yes in iax.conf.
2255 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2256 releases it was disabled.
2260 * The PBX core previously made a call with a non-existing extension test for
2261 extension s@default and jump there if the extension existed.
2262 This was a bad default behaviour and violated the principle of least surprise.
2263 It has therefore been changed in this release. It may affect some
2264 applications and configurations that rely on this behaviour. Most channel
2265 drivers have avoided this for many releases by testing whether the extension
2266 called exists before starting the PBX and generating a local error.
2267 This behaviour still exists and works as before.
2269 Extension "s" is used when no extension is given in a channel driver,
2270 like immediate answer in DAHDI or calling to a domain with no user part
2273 ------------------------------------------------------------------------------
2274 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2275 ------------------------------------------------------------------------------
2279 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2280 now defaults to force_rport. It is very important that phones requiring nat=no be
2281 specifically set as such instead of relying on the default setting. If at all
2282 possible, all devices should have nat settings configured in the general section as
2283 opposed to configuring nat per-device.
2284 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2285 codecs sent in response to an INVITE to the single most preferred codec.
2286 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2287 to be used for the outgoing call. It must be one of the codecs configured
2289 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2290 to be used for holding a private key. If tlsprivatekey is not specified,
2291 tlscertfile is searched for both public and private key.
2292 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2293 outbound client connections to be specified.
2294 * The sendrpid parameter has been expanded to include the options
2295 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2296 header to be sent (equivalent to setting sendrpid=yes) and setting
2297 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2298 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2299 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2300 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2301 will accept the SDP even if the SDP version number is not properly incremented,
2302 but will generate a warning in the log indicating that the SIP peer that sent
2303 the SDP should have the 'ignoresdpversion' option set.
2304 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2305 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2306 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2307 remote side requests it and disables symmetric RTP support. Setting it to
2308 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2309 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2310 and enables symmetric RTP support.
2311 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2312 response. This permits the master channel to know how each channel dialled
2313 in a multi-channel setup resolved in an individual way. This carries a
2314 performance penalty and can be disabled in sip.conf using the
2315 'storesipcause' option.
2316 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2317 configuration for the externip and externhost options when tcp or tls is used.
2318 * Added support for message body (stored in content variable) to SIP NOTIFY message
2319 accessible via AMI and CLI.
2320 * Added 'media_address' configuration option which can be used to explicitly specify
2321 the IP address to use in the SDP for media (audio, video, and text) streams.
2322 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2323 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2325 * Added 'use_q850_reason' configuration option for generating and parsing
2326 if available Reason: Q.850;cause=<cause code> header. It is implemented
2327 in some gateways for better passing PRI/SS7 cause codes via SIP.
2328 * When dialing SIP peers, a new component may be added to the end of the dialstring
2329 to indicate that a specific remote IP address or host should be used when dialing
2330 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2331 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2332 ability to selectively force bridged channels to also be encrypted is also
2333 implemented. Branching in the dialplan can be done based on whether or not
2334 a channel has secure media and/or signaling.
2335 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2337 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2338 Charge messages to snom phones.
2339 * Added support for G.719 media streams.
2340 * Added support for 16khz signed linear media streams.
2341 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2342 RTP has been outfitted with the same abilities.
2343 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2344 available in device configurations as well as in the dial plan.
2345 * Addition of the 'subscribe_network_change' option for turning on and off
2346 res_stun_monitor module support in chan_sip.
2347 * Addition of the 'auth_options_requests' option for turning on and off
2348 authentication for OPTIONS requests in chan_sip.
2352 * Add #tryinclude statement for config files. This provides the same
2353 functionality as the #include statement however an asterisk module will
2354 still load if the filename does not exist. Using the #include statement
2355 Asterisk will not allow the module to load.
2359 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2360 on realtime updates.
2361 * Added the ability for chan_iax2 to inform the dialplan whether or not
2362 encryption is being used. This interoperates with the SIP SRTP implementation
2363 so that a secure SIP call can be bridged to a secure IAX call when the
2364 dialplan requires bridged channels to be "secure".
2365 * Addition of the 'subscribe_network_change' option for turning on and off
2366 res_stun_monitor module support in chan_iax.
2371 * Added ability to preset channel variables on indicated lines with the setvar
2372 configuration option. Also, clearvars=all resets the list of variables back
2374 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2375 See configs/res_pktccops.conf for more information.
2377 XMPP Google Talk/Jingle changes
2378 -------------------------------
2379 * Added the externip option to gtalk.conf.
2380 * Added the stunaddr option to gtalk.conf which allows for the automatic
2381 retrieval of the external ip from a stun server.
2385 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2386 match to a partial channel name.
2387 * Added .m3u support for Mp3Player application.
2388 * Added progress option to the app_dial D() option. When progress DTMF is
2389 present, those values are sent immediately upon receiving a PROGRESS message
2390 regardless if the call has been answered or not.
2391 * Added functionality to the app_dial F() option to continue with execution
2392 at the current location when no parameters are provided.
2393 * Added the 'a' option to app_dial to answer the calling channel before any
2394 announcements or macros are executed.
2395 * Modified app_dial to set answertime when the called channel answers even if
2396 the called channel hangs up during playback of an announcement.
2397 * Modified app_dial 'r' option to support an additional parameter to play an
2398 indication tone from indications.conf
2399 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2400 to cycle through the next available channel. By default this is still '*'.
2401 * Added x() option to app_chanspy. This option allows DTMF to be set to
2402 exit the application.
2403 * The Voicemail application has been improved to automatically ignore messages
2404 that only contain silence.
2405 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2406 associated mailbox(es) to be greetings-only.
2407 * The ChanSpy application now has the 'S' option, which makes the application
2408 automatically exit once it hits a point where no more channels are available
2410 * The ChanSpy application also now has the 'E' option, which spies on a single
2411 channel and exits when that channel hangs up.
2412 * The MeetMe application now turns on the DENOISE() function by default, for
2413 each participant. In our tests, this has significantly decreased background
2414 noise (especially noisy data centers).
2415 * Voicemail now permits storage of secrets in a separate file, located in the
2416 spool directory of each individual user. The control for this is located in
2417 the "passwordlocation" option in voicemail.conf. Please see the sample
2418 configuration for more information.
2419 * The ChanIsAvail application now exposes the returned cause code using a separate
2420 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2421 * Added 'd' option to app_followme. This option disables the "Please hold"
2423 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2424 received will terminate recording.
2425 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2426 Previously the folder could only be set per context, but has now been extended
2427 using the imapfolder option.
2428 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2429 * Voicemail now allows the pager date format to be specified separately from the
2431 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2432 to allow joining, leaving, and sending text to group chats.
2433 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2434 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2435 to all paged phones (and optionally excluding the caller's one using the new
2436 option 'n') before the call is bridged.
2437 * The 'f' option to Dial has been augmented to take an optional argument. If no
2438 argument is provided, the 'f' option works as it always has. If an argument is
2439 provided, then the connected party information of all outgoing channels created
2440 during the Dial will be set to the argument passed to the 'f' option.
2441 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2443 * The OSP lookup application adds in/outbound network ID, optional security,
2444 number portability, QoS reporting, destination IP port, custom info and service
2446 * Added new application VMSayName that will play the recorded name of the voicemail
2447 user if it exists, otherwise will play the mailbox number.
2448 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2449 retrieve state for a particular bridge, where <name> is the conference name
2450 * app_directory now allows exiting at any time using the operator or pound key.
2451 * Voicemail now supports setting a locale per-mailbox.
2452 * Two new applications are provided for declining counting phrases in multiple
2453 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2455 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2457 * Voicemail now includes rdnis within msgXXXX.txt file.
2458 * ExternalIVR now supports IPv6 addresses.
2459 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2460 at https://wiki.asterisk.org/wiki/x/oQBB
2461 * ParkedCall and Park can now specify the parking lot to use.
2465 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2466 over SRV records associated with a specific service. From the CLI, type
2467 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2468 details on how these may be used.
2469 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2470 pitch of a channel's tx and rx audio streams.
2471 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2472 setting various connected line and redirecting party information.
2473 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2474 support ISDN subaddressing.
2475 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2476 * For DAHDI channels, the CHANNEL() dialplan function now allows
2477 the dialplan to request changes in the configuration of the active
2478 echo canceller on the channel (if any), for the current call only.
2481 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2483 The possible values are:
2485 on - normal mode (the echo canceller is actually reinitialized)
2487 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2489 voice - voice mode (returns from FAX mode, reverting the changes that
2490 were made when FAX mode was requested)
2491 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2492 and setting variables on the channel which created the current channel.
2493 Administrators should take care to avoid naming conflicts, when multiple
2494 channels are dialled at once, especially when used with the Local channel
2495 construct (which all could set variables on the master channel). Usage
2496 of the HASH() dialplan function, with the key set to the name of the slave
2497 channel, is one approach that will avoid conflicts.
2498 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2500 * func_odbc now allows multiple row results to be retrieved without using
2501 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2502 from the same query by using the name of the function which retrieved the
2503 first row as an argument to ODBC_FETCH().
2504 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2505 dialplan. This function returns the content of the received message.
2506 * Added REPLACE, which searches a given variable name for a set of characters,
2507 then either replaces them with a single character or deletes them.
2508 * Added PASSTHRU, which literally passes the same argument back as its return
2509 value. The intent is to be able to use a literal string argument to
2510 functions that currently require a variable name as an argument.
2511 * HASH-associated variables now can be inherited across channel creation, by
2512 prefixing the name of the hash at assignment with the appropriate number of
2513 underscores, just like variables.
2514 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2515 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2516 whether or not channels that are bridged to the current channel will be
2517 required to have secure signaling and/or media.
2518 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2519 the current channel has secure signaling and/or media.
2520 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2521 "no_media_path" option.
2522 Returns "0" if there is a B channel associated with the call.
2523 Returns "1" if no B channel is associated with the call. The call is either
2524 on hold or is a call waiting call.
2525 * Added option to dialplan function CDR(), the 'f' option
2526 allows for high resolution times for billsec and duration fields.
2527 * FILE() now supports line-mode and writing.
2528 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2529 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2533 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2534 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2535 and is set when a dynamic feature is triggered.
2536 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2537 to dynamically create a new parking lot matching the value this varible is
2539 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2540 features.conf that should be the base for dynamic parkinglots.
2541 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2542 parkinglot should have.
2543 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2544 parkinglot should have.
2545 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2550 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2551 timeout has expired.
2552 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2553 to the caller when an Agent's phone is ringing. This can be used to indicate
2554 to the caller that their call is about to be picked up, which is nice when
2555 one has been on hold for an extened period of time.
2556 * A new config option, penaltymemberslimit, has been added to queues.conf.
2557 When set this option will disregard penalty settings when a queue has too
2559 * A new option, 'I' has been added to both app_queue and app_dial.
2560 By setting this option, Asterisk will not update the caller with
2561 connected line changes or redirecting party changes when they occur.
2562 * A 'relative-periodic-announce' option has been added to queues.conf. When
2563 enabled, this option will cause periodic announce times to be calculated
2564 from the end of announcements rather than from the beginning.
2565 * The autopause option in queues.conf can be passed a new value, "all." The
2566 result is that if a member becomes auto-paused, he will be paused in all
2567 queues for which he is a member, not just the queue that failed to reach
2569 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2570 * The queue logger now allows events to optionally propagate to a file,
2571 even when realtime logging is turned on. Additionally, realtime logging
2572 supports sending the event arguments to 5 individual fields, although it
2573 will fallback to the previous data definition, if the new table layout is
2576 mISDN channel driver (chan_misdn) changes
2577 ----------------------------------------
2578 * Added display_connected parameter to misdn.conf to put a display string
2579 in the CONNECT message containing the connected name and/or number if
2580 the presentation setting permits it.
2581 * Added display_setup parameter to misdn.conf to put a display string
2582 in the SETUP message containing the caller name and/or number if the
2583 presentation setting permits it.
2584 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2585 indicate the dialplan settings are to be obtained from the asterisk
2587 * Made misdn.conf parameter callerid accept the "name" <number> format
2588 used by the rest of the system.
2589 * Made use the nationalprefix and internationalprefix misdn.conf
2590 parameters to prefix any received number from the ISDN link if that
2591 number has the corresponding Type-Of-Number. NOTE: This includes
2592 comparing the incoming call's dialed number against the MSN list.
2593 * Added the following new parameters: unknownprefix, netspecificprefix,
2594 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2595 received number from the ISDN link if that number has the corresponding
2597 * Added new dialplan application misdn_command which permits controlling
2598 the CCBS/CCNR functionality.
2599 * Added new dialplan function mISDN_CC which permits retrieval of various
2600 values from an active call completion record.
2601 * For PTP, you should manually send the COLR of the redirected-to party
2602 for an incomming redirected call if the incoming call could experience
2603 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2604 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2605 if the REDIRECTING(from-num) is not empty.
2606 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2607 option on all of the REDIRECTING statements before dialing the
2608 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2609 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2610 redirecting-to presentation (COLR) when it becomes available.
2611 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2614 thirdparty mISDN enhancements
2615 -----------------------------
2616 mISDN has been modified by Digium, Inc. to greatly expand facility message
2618 * Enhanced COLP support for call diversion and transfer.
2619 * CCBS/CCNR support.
2621 The latest modified mISDN v1.1.x based version is available at:
2622 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2623 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2625 Tagged versions of the modified mISDN code are available under:
2626 http://svn.digium.com/svn/thirdparty/mISDN/tags
2627 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2629 libpri channel driver (chan_dahdi) DAHDI changes
2630 -------------------------------------------
2631 * The channel variable PRIREDIRECTREASON is now just a status variable
2632 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2633 to read and alter the reason.
2634 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2635 redirected-to party for an incomming redirected call if the incoming call
2636 could experience further redirects. Just set the
2637 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2638 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2640 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2641 use the inhibit(i) option on all of the REDIRECTING statements before
2642 dialing the redirected-to party. You still have to set the
2643 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2644 will update the redirecting-to presentation (COLR) when it becomes available.
2645 * Added the ability to ignore calls that are not in a Multiple Subscriber
2646 Number (MSN) list for PTMP CPE interfaces.
2647 * Added dynamic range compression support for dahdi channels. It is
2648 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2649 * Added support for ISDN calling and called subaddress with partial support
2650 for connected line subaddress.
2651 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2652 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2653 to transfer a held call on disconnect similar to an analog phone.
2654 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2655 Will reroute/deflect an outgoing call when receive the message.
2656 Can use the DAHDISendCallreroutingFacility to send the message for the
2658 * Added standard location to add options to chan_dahdi dialing:
2659 Dial(DAHDI/g1[/extension[/options]])
2662 R Reverse charging indication
2663 * Added Reverse Charging Indication (Collect calls) send/receive option.
2664 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2665 Dial(DAHDI/g1/extension/R)
2666 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2667 (requires latest LibPRI)
2668 * Added ability to send/receive keypad digits in the SETUP message.
2669 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2670 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2671 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2672 (requires latest LibPRI)
2673 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2674 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2675 back into the same interface. Tromboned calls happen because of call routing,
2676 call deflection, call forwarding, and call transfer.
2677 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2678 * Added the ability to support call waiting calls. (The SETUP has no B channel
2680 * Added Malicious Call ID (MCID) event to the AMI call event class.
2681 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2683 Asterisk Manager Interface
2684 --------------------------
2685 * The Hangup action now accepts a Cause header which may be used to
2686 set the channel's hangup cause.
2687 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2688 to specify a separate .pem file to hold a private key. By default sslcert
2689 is used to hold both the public and private key.
2690 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2691 for options containing the 'tls' prefix. For example, 'sslenable' is now
2692 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2693 across all .conf files. All affected sample.conf files have been modified to
2694 reflect this change. Previous options such as 'sslenable' still work,
2695 but options with the 'tls' prefix are preferred.
2696 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2697 in a channel. (res_mutestream.so)
2698 * The configuration file manager.conf now supports a channelvars option, which
2699 specifies a list of channel variables to include in each channel-oriented
2701 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2702 and ExtraPriority to allow redirecting the second channel to a different
2703 location than the first.
2704 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2706 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2707 in a MixMonitor recording.
2708 * The 'iax2 show peers' output is now similar to the expected output of
2710 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2712 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2713 AOC-E messages on a channel.
2714 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2715 conform more closely to similar events.
2716 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2718 * Added optional parkinglot variable for park command.
2719 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2720 if CallerIDNum and CallerIDName headers are also present.
2722 Channel Event Logging
2723 ---------------------
2724 * A new interface, CEL, is introduced here. CEL logs single events, much like
2725 the AMI, but it differs from the AMI in that it logs to db backends much
2726 like CDR does; is based on the event subsystem introduced by Russell, and
2727 can share in all its benefits; allows multiple backends to operate like CDR;
2728 is specialized to event data that would be of concern to billing sytems,
2729 like CDR. Backends for logging and accounting calls have been produced,
2730 but a new CDR backend is still in development.
2734 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2735 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2736 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2737 * Multiple files and formats can now be specified in cdr_custom.conf.
2738 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2739 See configs/cdr_syslog.conf.sample for more information.
2740 * A 'sequence' field has been added to CDRs which can be combined with
2741 linkedid or uniqueid to uniquely identify a CDR.
2742 * Handling of billsec and duration field has changed. If your table definition
2743 specifies those fields as float,double or similar they will now be logged with
2744 microsecond accuracy instead of a whole integer.
2746 Calendaring for Asterisk
2747 ------------------------
2748 * A new set of modules were added supporing calendar integration with Asterisk.
2749 Dialplan functions for reading from and writing to calendars are included,
2750 as well as the ability to execute dialplan logic upon calendar event notifications.
2751 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2752 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2753 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2754 2003 support does not support forms-based authentication).
2756 Call Completion Supplementary Services for Asterisk
2757 ---------------------------------------------------
2758 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2759 DAHDI/ISDN supports call completion for the following switch types:
2760 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2761 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2763 Multicast RTP Support
2764 ---------------------
2765 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2766 The channel driver can be used with the Page application to perform multicast RTP
2767 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2768 Type can be either basic or linksys.
2769 Destination is the IP address and port for the RTP packets.
2770 Control address is specific to the linksys type and is used for sending the control
2771 packets unique to them.
2773 Security Events Framework
2774 -------------------------
2775 * Asterisk has a new C API for reporting security events. The module res_security_log
2776 sends these events to the "security" logger level. Currently, AMI is the only
2777 Asterisk component that reports security events. However, SIP support will be
2778 coming soon. For more information on the security events framework, see the
2779 "Asterisk Security Framework" section of the Asterisk wiki at
2780 https://wiki.asterisk.org/wiki/x/wgBQ
2781 * SIP support was added in Asterisk 10
2782 * This API now supports IPv6 addresses
2786 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2787 * A spandsp based fax backend (res_fax_spandsp) has been added.
2788 * The app_fax module has been deprecated in favor of the res_fax module and
2789 the new res_fax_spandsp backend.
2790 * The SendFAX and ReceiveFAX applications now send their log messages to a
2791 'fax' logger level, instead of to the generic logger levels. To see these
2792 messages, the system's logger.conf file will need to direct the 'fax' logger
2793 level to one or more destinations; the logger.conf.sample file includes an
2794 example of how to do this. Note that if the 'fax' logger level is *not*
2795 directed to at least one destination, log messages generated by these
2796 applications will be lost, and that if the 'fax' logger level is directed to
2797 the console, the 'core set verbose' and 'core set debug' CLI commands will
2798 have no effect on whether the messages appear on the console or not.
2802 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2803 Now, in order to enable transmitting silence during record the transmit_silence
2804 option should be used. transmit_silence_during_record remains a valid option, but
2805 defaults to the behavior of the transmit_silence option.
2806 * Addition of the Unit Test Framework API for managing registration and execution
2807 of unit tests with the purpose of verifying the operation of C functions.
2808 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2809 XMPP text messages to the remote JID.
2810 * Modules.conf has a new option - "require" - that marks a module as critical for
2811 the execution of Asterisk.
2812 If one of the required modules fail to load, Asterisk will exit with a return
2814 * An 'X' option has been added to the asterisk application which enables #exec support.
2815 This allows #exec to be used in asterisk.conf.
2816 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2817 * A new lockconfdir option has been added to asterisk.conf to protect the
2818 configuration directory (/etc/asterisk by default) during reloads.
2819 * The parkeddynamic option has been added to features.conf to enable the creation
2820 of dynamic parkinglots.
2821 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2822 the reportalarms config option.
2823 * chan_dahdi supports dialing configuring and dialing by device file name.
2824 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2825 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2826 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2827 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2828 Handy for the above name-based syntax as it does not depend on
2829 initialization order.
2830 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2831 significant increase in performance (about 3X) for installations using this switchtype.
2832 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2833 AIS. For more information, please see the Distributed Device State section of the
2834 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2835 * The addition of G.719 pass-through support.
2836 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2837 during device configuration.
2838 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2839 have less than 3 lines on the LCD.
2840 * Realtime now supports database failover. See the sample extconfig.conf for details.
2841 * The addition of improved translation path building for wideband codecs. Sample
2842 rate changes during translation are now avoided unless absolutely necessary.
2843 * The addition of the res_stun_monitor module for monitoring and reacting to network
2844 changes while behind a NAT.
2845 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2846 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2847 These allow support for any Administration. Default is AT&T values.
2851 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2852 optionally accept a filename, to apply the setting only to the code generated from
2853 that source file when Asterisk was built. However, there are some modules in Asterisk
2854 that are composed of multiple source files, so this did not result in the behavior
2855 that users expected. In this version, 'core set debug' and 'core set verbose'
2856 can optionally accept *module* names instead (with or without the .so extension),
2857 which applies the setting to the entire module specified, regardless of which source
2858 files it was built from.
2859 * New 'manager show settings' command showing the current settings loaded from
2861 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2862 the channel hangup request to all channels.
2863 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2865 ------------------------------------------------------------------------------
2866 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2867 ------------------------------------------------------------------------------
2871 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2872 Snom phones use this for call pickup of extensions that the phone is
2874 * Added support for setting the domain in the URI for caller of an
2875 outbound call by using the SIPFROMDOMAIN channel variable.
2876 * Added a new configuration option "remotesecret" for authentication to
2877 remote services. For backwards compatibility, "secret" still has the
2878 same function as before, but now you can configure both a remote secret and a
2879 local secret for mutual authentication.
2880 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2881 the sound will be played to the target of an attended transfer
2882 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2883 finer control over how many peers Asterisk will qualify and the gap between them
2884 when all peers need to be qualified at the same time.
2885 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2886 (either globally or for a specific peer), chan_sip will treat any SDP data
2887 it receives as new data and update the media stream accordingly. By
2888 default, Asterisk will only modify the media stream if the SDP session
2889 version received is different from the current SDP session version. This
2890 option is required to interoperate with devices that have non-standard SDP
2891 session version implementations (observed with Microsoft OCS). This option
2892 is disabled by default.
2893 * The parsing of register => lines in sip.conf has been modified to allow a port
2894 to be present in the "user" portion. Please see the sip.conf.sample file for more
2896 * Added support for subscribing to MWI on a remote server and making the status available
2897 as a mailbox. Please see the sip.conf.sample file for more information.
2898 * Added a function to remove SIP headers added in the dialplan before the
2899 first INVITE is generated - SIPRemoveHeader()
2900 * Channel variables set with setvar= in a device configuration is now
2901 set both for inbound and outbound calls.
2902 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2906 * Added immediate option to iax.conf
2907 * Added forceencryption option to iax.conf
2908 * Added Encryption and Trunk status to manager command "iaxpeers"
2912 * The configuration file now holds separate sections for devices and lines.
2913 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2918 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2919 support for LibOpenR2. http://www.libopenr2.org/
2920 * The UK option waitfordialtone has been added for use with BT analog
2922 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2923 is used in conjunction with the 'faxdetect' configuration option. When
2924 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2925 switch to the configured faxbuffers policy. For example, to use 6 buffers
2926 and a 'full' buffer policy for a fax transmission, add:
2928 The faxbuffers configuration will be in affect until the call is torn down.
2929 * Added service message support for 4ESS/5ESS switches.
2933 * For DAHDI channels, the CHANNEL() dialplan function now
2934 supports changing the channel's buffer policy (for the current
2935 call only), using this syntax:
2937 exten => s,n,Set(CHANNEL(buffers)=6,full)
2939 This would change the channel to the 'full' buffer policy and
2940 6 (six) buffers. Possible options for this setting are the same
2941 as those in chan_dahdi.conf.
2942 * Added a new dialplan function, CURLOPT, which permits setting various
2943 options that may be useful with the CURL dialplan function, such as
2944 cookies, proxies, connection timeouts, passwords, etc.
2945 * Permit the syntax and synopsis fields of the corresponding dialplan
2946 functions to be individually set from func_odbc.conf.
2947 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2948 * func_odbc now may specify an insert query to execute, when the write query
2949 affects 0 rows (usually indicating that no such row exists).
2950 * Added a new dialplan function, LISTFILTER, which permits removing elements
2951 from a set list, by name. Uses the same general syntax as the existing CUT
2952 and FIELDQTY dialplan functions, which also manage lists.
2953 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2954 obtaining realtime data from the dialplan.
2955 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2956 a subroutine when using the GoSub() and Return() applications.
2957 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2958 of "core show function AUDIOHOOK_INHERIT" from the CLI
2959 * Added AES_ENCRYPT. For information on its use, please see the output
2960 of "core show function AES_ENCRYPT" from the CLI
2961 * Added AES_DECRYPT. For information on its use, please see the output
2962 of "core show function AES_DECRYPT" from the CLI
2963 * func_odbc now supports database transactions across multiple queries.
2967 * Scheduled meetme conferences may now have their end times extended by
2969 * app_authenticate now gives the ability to select a prompt other than
2971 * app_directory now pays attention to the searchcontexts setting in
2972 voicemail.conf and will look through all contexts, if no context is
2973 specified in the initial argument.
2974 * A new application, Originate, has been introduced, that allows asynchronous
2975 call origination from the dialplan.
2976 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2977 in addition to the setting in the "general" context.
2978 * Added ConfBridge dialplan application which does conference bridges without
2979 DAHDI. For information on its use, please see the output of
2980 "core show application ConfBridge" from the CLI.
2984 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2985 operation to the AMI Redirect action.
2986 * extensions.conf now allows you to use keyword "same" to define an extension
2987 without actually specifying an extension. It uses exactly the same pattern
2988 as previously used on the last "exten" line. For example:
2989 exten => 123,1,NoOp(something)
2990 same => n,SomethingElse()
2991 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2992 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2993 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2994 by the new clialiases module. See cli_aliases.conf.sample file.
2995 * Times within timespecs are now accurate down to the minute. This is a change
2996 from historical Asterisk, which only provided timespecs rounded to the nearest
2997 even (read: evenly divisible by 2) minute mark.
2998 * The realtime switch now supports an option flag, 'p', which disables searches for
3000 * In addition to a time range and date range, timespecs now accept a 5th optional
3001 argument, timezone. This allows you to perform time checks on alternate
3002 timezones, especially if those daylight savings time ranges vary from your
3003 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3005 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3006 give you the correct output for an asterisk box behind nat. It will give you the
3007 externhost and localnet settings.
3008 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3009 can connect calls in passthrough mode, as well as record and play back files.
3010 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3011 using pickupsound and pickupfailsound in features.conf.
3012 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3013 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3014 instead of the /var/run/asterisk.pid where it used to be. This will make
3015 installs as non-root easier to manage.
3020 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3021 be written; they will no longer be explicitly written.
3023 Asterisk Manager Interface
3024 --------------------------
3025 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3026 a non-empty value) in your request. If you do this, any pending AMI events will
3027 *not* be included in the response to your request as they would normally, but
3028 will be left in the event queue for the next request you make to retrieve. For
3029 some applications, this will allow you to guarantee that you will only see
3030 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3031 To know whether the Asterisk server supports this header or not, your client can
3032 inspect the first response back from the server to see if it includes this header:
3034 Pragma: SuppressEvents
3036 If this is included, the server supports event suppression.
3038 * Added 4 new Actions to list skinny device(s) and line(s)
3044 LDAP Schema File Additions
3045 --------------------------
3046 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3047 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3049 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3050 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3051 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3052 * Removed redundant IPaddr (there's already IPAddress)
3053 - Gives more configuration Flags for SIP-Users available (tested)
3054 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3055 without extensibleObject (which really should be the last resort); gives
3056 also additional possibilities for LDAP-filter
3058 ------------------------------------------------------------------------------
3059 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3060 ------------------------------------------------------------------------------
3062 Device State Handling
3063 ---------------------
3064 * The event infrastructure in Asterisk got another big update to help support
3065 distributed events. It currently supports distributed device state and
3066 distributed Voicemail MWI (Message Waiting Indication). A new module has
3067 been merged, res_ais, which facilitates communicating events between servers.
3068 It uses the SAForum AIS (Service Availability Forum Application Interface
3069 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3070 a cluster of Asterisk servers, and to share events between them. For more
3071 information on setting this up, refer to the Distributed Device State section
3072 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3076 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3077 variables from an Asterisk configuration file.
3078 * The JACK_HOOK function now has a c() option to supply a custom client name.
3079 * Added two new dialplan functions from libspeex for audio gain control and
3080 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3081 rx directions of a channel from the dialplan.
3082 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3083 based on other parameters. The default is still to search based on the
3084 forwarding station ID. However, there are new options that allow you to search
3085 based on the message desk terminal ID, or the message desk number.
3086 * TIMEOUT() has been modified to be accurate down to the millisecond.
3087 * ENUM*() functions now include the following new options:
3088 - 'u' returns the full URI and does not strip off the URI-scheme.
3089 - 's' triggers ISN specific rewriting
3090 - 'i' looks for branches into an Infrastructure ENUM tree
3091 - 'd' for a direct DNS lookup without any flipping of digits.
3092 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3093 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3094 deviation of jitter, rtt, and loss for a call using chan_sip.
3096 DAHDI channel driver (chan_dahdi) Changes
3097 ----------------------------------------
3098 * Channels can now be configured using named sections in chan_dahdi.conf, just
3099 like other channel drivers, including the use of templates.
3100 * The default for pridialplan has changed from 'national' to 'unknown'.
3104 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3105 to something that matches the pattern a hint will be created using the contents
3106 and variables evaluated.
3107 * Dialplan matching has been extended to allow an extension to return to the
3108 PBX core to wait for more digits. This is done by using the new dialplan
3109 application called "Incomplete". This will permit a whole new level of
3110 extension control, by giving the administrator more control over early
3111 matches employing one of the short-circuit pattern match operators. Note
3112 that custom applications can trigger this same behavior by returning the
3113 special value AST_PBX_INCOMPLETE.
3117 * Directory now permits both first and last names to be matched at the same
3118 time. In addition, the number of digits to enter of the name can be set in
3119 the arguments to Directory; previously, you could enter only 3, regardless
3120 of how many names are in your company. For large companies, this should be
3122 * Voicemail now permits a mailbox setting to wrap around from first to last
3123 messages, if the "messagewrap" option is set to a true value.
3124 * Voicemail now permits an external script to be run, for password validation.
3125 The script should output "VALID" or "INVALID" on stdout, depending upon the
3126 wish to validate or invalidate the password given. Arguments are:
3127 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3129 * Dial has a new option: F(context^extension^pri), which permits a callee to
3130 continue in the dialplan, at the specified label, if the caller hangs up.
3131 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3132 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3133 * The Jack application now has a c() option to supply a custom client name.
3134 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3135 like the pre-existing whisper mode, except that the spy can also talk to the
3136 participant on the bridged channel as well.
3137 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3138 to be spoken instead of the channel name or number. For more information on the
3139 use of this option, issue the command "core show application ChanSpy" from the
3141 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3142 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3143 words, if using the 'd' option, it is not possible to enter a number to append to
3144 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3145 change to whisper mode, and pressing 6 will change to barge mode.
3146 * ExternalIVR now takes several options that affect the way it performs, as
3147 well as having several new commands. Please see the External IVR page on the Asterisk
3148 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3149 * Added ability to communicate over a TCP socket instead of forking a child process for the
3150 ExternalIVR application.
3151 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3152 of just the first one if you give the function more then one channel to check.
3153 * PrivacyManager now takes an option where you can specify a context where the
3154 given number will be matched. This way you have more control over who is allowed
3155 and it stops the people who blindly enter 10 digits.
3156 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3157 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3158 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3159 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3160 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3161 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3162 * The Dial() application no longer copies the language used by the caller to the callee's
3163 channel. If you desire for the caller's channel's language to be used for file playback
3164 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3165 * SendImage() no longer hangs up the channel on error; instead, it sets the
3166 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3167 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3169 * Park has a new option, 's', which silences the announcement of the parking space number.
3170 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3171 invalid input and will be assumed to mean that no timeout is desired.
3175 * Added DNS manager support to registrations for peers referencing peer entries.
3176 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3177 as well as periodically updating the IP address. These properties allow for
3178 better performance as well as recovery in the event of an IP change.
3179 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3180 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3181 These changes also provide performance improvements for call setup and tear down.
3182 * Added ability to specify registration expiry time on a per registration basis in
3184 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3186 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3187 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3188 * 'sip show peers' and 'sip show users' display their entries sorted in
3189 alphabetical order, as opposed to the order they were in, in the config
3191 * Videosupport now supports an additional option, "always", which always sets
3192 up video RTP ports, even on clients that don't support it. This helps with
3193 callfiles and certain transfers to ensure that if two video phones are
3194 connected, they will always share video feeds.
3198 * Existing DNS manager lookups extended to check for SRV records.
3199 * IAX2 encryption support has been improved to support periodic key rotation
3200 within a call for enhanced security. The option "keyrotate" has been
3201 provided to disable this functionality to preserve backwards compatibility
3202 with older versions of IAX2 that do not support key rotation.
3206 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3207 data tree based on the given <path>.
3208 * New CLI command "data show providers" that will display all the registered
3210 * New CLI command, "config reload <file.conf>" which reloads any module that
3211 references that particular configuration file. Also added "config list"
3212 which shows which configuration files are in use.
3213 * New CLI commands, "pri show version" and "ss7 show version" that will
3214 display which version of libpri and libss7 are being used, respectively.
3215 A new API call was added so trunk will now have to be compiled against
3216 a versions of libpri and libss7 that have them or it will not know that
3217 these libraries exist.
3218 * The commands "core show globals", "core set global" and "core set chanvar" has
3219 been deprecated in favor of the more semanticly correct "dialplan show globals",
3220 "dialplan set chanvar" and "dialplan set global".
3221 * New CLI command "dialplan show chanvar" to list all variables associated
3222 with a given channel.
3226 * Addresses managed by DNS manager now can check to see if there is a DNS
3227 SRV record for a given domain and will use that hostname/port if present.
3229 AMI - The manager (TCP/TLS/HTTP)
3230 --------------------------------
3231 * The Status command now takes an optional list of variables to display
3232 along with channel status.
3233 * The QueueEntry event now also includes the channel's uniqueid
3237 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3238 as some people were running into this limit. This limit has been increased
3243 * The TRANSFER queue log entry now includes the the caller's original
3244 position in the transferred-from queue.
3245 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3246 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3247 as well as an explanation about timeout options in general
3248 * Added a new option - C - for forcing the "answered elsewhere" flag on
3249 cancellation of calls in to members of the queue. This is to avoid the
3250 call to a member of a queue having the call listed as a "missed call".
3254 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3255 adaptive capabilities. What this means in practical terms is that if your
3256 realtime table lacks critical fields, Asterisk will now emit warnings to
3257 that effect. Also, some of the realtime drivers have the ability (if
3258 configured) to automatically add those columns to the table with the
3259 correct type and length.
3263 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3264 the 'setvar' option to cause a given audio file to be played upon completion
3265 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3266 Skinny channels only.
3267 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3268 Hoard page on the Asterisk wiki for more information:
3269 https://wiki.asterisk.org/wiki/x/pQBB
3270 * Config file variables may now be appended to, by using the '+=' append
3271 operator. This is most helpful when working with long SQL queries in
3272 func_odbc.conf, as the queries no longer need to be specified on a single
3274 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3275 which will add a second to the billsec when the ending
3276 time is set, if the number in the microseconds field of the end time is
3277 greater than the number of microseconds in the answer time. This allows
3278 users to count the 'initiated' seconds in their billing records.
3280 ------------------------------------------------------------------------------
3281 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3282 ------------------------------------------------------------------------------
3284 AMI - The manager (TCP/TLS/HTTP)
3285 --------------------------------
3286 * Manager has undergone a lot of changes, all of them documented
3287 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3288 * Manager version has changed to 1.1
3289 * Added a new action 'CoreShowChannels' to list currently defined channels
3290 and some information about them.
3291 * Added a new action 'SIPshowregistry' to list SIP registrations.
3292 * Added TLS support for the manager interface and HTTP server
3293 * Added the URI redirect option for the built-in HTTP server
3294 * The output of CallerID in Manager events is now more consistent.
3295 CallerIDNum is used for number and CallerIDName for name.
3296 * Enable https support for builtin web server.
3297 See configs/http.conf.sample for details.
3298 * Added a new action, GetConfigJSON, which can return the contents of an
3299 Asterisk configuration file in JSON format. This is intended to help
3300 improve the performance of AJAX applications using the manager interface
3302 * SIP and IAX manager events now use "ChannelType" in all cases where we
3303 indicate channel driver. Previously, we used a mixture of "Channel"
3304 and "ChannelDriver" headers.
3305 * Added a "Bridge" action which allows you to bridge any two channels that
3306 are currently active on the system.
3307 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3308 the voicemail users setup.
3309 * Added 'DBDel' and 'DBDelTree' manager commands.
3310 * cdr_manager now reports events via the "cdr" level, separating it from
3311 the very verbose "call" level.
3312 * Manager users are now stored in memory. If you change the manager account
3313 list (delete or add accounts) you need to reload manager.
3314 * Added Masquerade manager event for when a masquerade happens between
3316 * Added "manager reload" command for the CLI
3317 * Lots of commands that only provided information are now allowed under the
3318 Reporting privilege, instead of only under Call or System.
3319 * The IAX* commands now require either System or Reporting privilege, to
3320 mirror the privileges of the SIP* commands.
3321 * Added ability to retrieve list of categories in a config file.
3322 * Added ability to retrieve the content of a particular category.
3323 * Added ability to empty a context.
3324 * Created new action to create a new file.
3325 * Updated delete action to allow deletion by line number with respect to category.
3326 * Added new action insert to add new variable to category at specified line.
3327 * Updated action newcat to allow new category to be inserted in file above another
3329 * Added new event "JitterBufStats" in the IAX2 channel
3330 * Originate now requires the Originate privilege and, if you want to call out
3331 to a subshell, it requires the System privilege, as well. This was done to
3332 enhance manager security.
3333 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3334 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3335 or manager show command Atxfer from the CLI
3336 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3337 details or manager show command IAXregistry from the CLI
3341 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3342 state in the dialplan, as well as creating custom device states that are
3343 controllable from the dialplan.
3344 * Extend CALLERID() function with "pres" and "ton" parameters to
3345 fetch string representation of calling number presentation indicator
3346 and numeric representation of type of calling number value.
3347 * MailboxExists converted to dialplan function
3348 * A new option to Dial() for telling IP phones not to count the call
3349 as "missed" when dial times out and cancels.
3350 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3351 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3352 held for any given channel. Also, locks are automatically freed when a
3354 * Added HINT() dialplan function that allows retrieving hint information.
3355 Hints are mappings between extensions and devices for the sake of
3356 determining the state of an extension. This function can retrieve the list
3357 of devices or the name associated with a hint.
3358 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3360 * Added SYSINFO() dialplan function which allows retrieval of system information
3361 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3362 the existence of a dialplan target.
3363 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3364 upper and lower case, respectively.
3365 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3366 ID for the call (not the Asterisk call ID or unique ID), provided that the
3367 channel driver supports this. For SIP, you get the SIP call-ID for the
3368 bridged channel which you can store in the CDR with a custom field.
3372 * Added CLI permissions, config file: cli_permissions.conf
3373 default is to allow all commands for every local user/group.
3374 Also this new feature added three new CLI commands:
3375 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3376 - cli reload permissions
3377 - cli show permissions
3378 * New CLI command "core show hint" (usage: core show hint <exten>)
3379 * New CLI command "core show settings"
3380 * Added 'core show channels count' CLI command.
3381 * Added the ability to set the core debug and verbose values on a per-file basis.
3382 * Added 'queue pause member' and 'queue unpause member' CLI commands
3383 * Ability to set process limits ("ulimit") without restarting Asterisk
3384 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3385 output to make debugging on busy systems much easier.
3386 * New CLI commands "dialplan set extenpatternmatching true/false"
3387 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3388 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3389 listed in the startup_commands section of cli.conf will get executed.
3390 * Added a CLI command, "devstate change", which allows you to set custom device
3391 states from the func_devstate module that provides the DEVICE_STATE() function
3392 and handling of the "Custom:" devices.
3393 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3394 sorted into the different possible callbacks, with the number of entries
3395 currently scheduled for each. Gives you a feel for how busy the sip channel
3397 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3398 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3399 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3403 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3404 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3405 for a received call. If it is detected, the channel will jump to the
3406 'fax' extension in the dialplan.
3407 * The default SIP useragent= identifier now includes the Asterisk version
3408 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3409 If set, and the incoming request carries authentication info,
3410 the username to match in the users list is taken from the Digest header
3411 rather than from the From: field. This feature is considered experimental.
3412 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3413 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3414 * The "localmask" setting was removed in version 1.2 and the reminder about it
3415 being removed is now also removed.
3416 * A new option "busylevel" for setting a level of calls where asterisk reports
3417 a device as busy, to separate it from call-limit. This value is also added
3418 to the SIP_PEER dialplan function.
3419 * A new realtime family called "sipregs" is now supported to store SIP registration
3420 data. If this family is defined, "sippeers" will be used for configuration and
3421 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3422 registration data, as before.
3423 * The SIPPEER function have new options for port address, call and pickup groups
3424 * Added support for T.140 realtime text in SIP/RTP
3425 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3426 required due to the restructuring of how MWI is handled. See the descriptions
3427 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3428 for more information.
3429 * Added rtpdest option to CHANNEL() dialplan function.
3430 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3431 * SIP now adds a header to the CANCEL if the call was answered by another phone
3432 in the same dial command, or if the new c option in dial() is used.
3433 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3434 states it is not needed. For phones, however, that do require it the "registertrying" option
3435 has been added so it can be enabled.
3436 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3437 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3438 used to enable this functionality).
3439 * New settings for timer T1 and timer B on a global level or per device. This makes it
3440 possible to force timeout faster on non-responsive SIP servers. These settings are
3441 considered advanced, so don't use them unless you have a problem.
3442 * Added a dial string option to be able to set the To: header in an INVITE to any
3444 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3445 the qualify frequency.
3446 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3447 were not properly torn down due to network or endpoint failures during an established
3449 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3450 and configs/sip.conf.sample for more information on how it is used.
3451 * Added a new configuration option "authfailureevents" that enables manager events when
3452 a peer can't authenticate properly.
3453 * Added DNS manager support to registrations for peers not referencing a peer entry.
3457 * Added the trunkmaxsize configuration option to chan_iax2.
3458 * Added the srvlookup option to iax.conf
3459 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3462 XMPP Google Talk/Jingle changes
3463 -------------------------------
3464 * Added the bindaddr option to gtalk.conf.
3468 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
3469 * Proper codec support in chan_skinny.
3470 * Added settings for IP and Ethernet QoS requests
3474 * Added separate settings for media QoS in mgcp.conf
3476 Console Channel Driver changes
3477 ------------------------------
3478 * Added experimental support for video send & receive to chan_oss.
3479 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
3482 Phone channel changes (chan_phone)
3483 ----------------------------------
3484 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
3486 H.323 channel Changes
3487 ---------------------
3488 * H323 remote hold notification support added (by NOTIFY message
3489 and/or H.450 supplementary service)
3491 Local channel changes
3492 ---------------------
3493 * The device state functionality in the Local channel driver has been updated
3494 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
3495 to just UNKNOWN if the extension exists.
3496 * Added jitterbuffer support for chan_local. This allows you to use the
3497 generic jitterbuffer on incoming calls going to Asterisk applications.
3498 For example, this would allow you to use a jitterbuffer for an incoming
3499 SIP call to Voicemail by putting a Local channel in the middle. This
3500 feature is enabled by using the 'j' option in the Dial string to the Local
3501 channel in conjunction with the existing 'n' option for local channels.
3502 * A 'b' option has been added which causes chan_local to return the actual channel
3503 that is behind it when queried. This is useful for transfer scenarios as the
3504 actual channel will be transferred, not the Local channel.
3506 Agent channel changes
3507 ----------------------
3508 * The ackcall and endcall options are now supplemented with options acceptdtmf
3509 and enddtmf. These allow for the DTMF keypress to be configurable. The options
3510 default to their old hard-coded values ('#' and '*' respectively) so this should
3511 not break any existing agent installations.
3513 DAHDI channel driver (chan_dahdi) Changes
3514 ----------------------------------------
3515 * SS7 support (via libss7 library)
3516 * In India, some carriers transmit CID via dtmf. Some code has been added
3517 that will handle some situations. The cidstart=polarity_IN choice has been added for
3518 those carriers that transmit CID via dtmf after a polarity change.
3519 * CID matching information is now shown when doing 'dialplan show'.
3520 * Added dahdi show version CLI command.
3521 * Added setvar support to chan_dahdi.conf channel entries.
3522 * Added two new options: mwimonitor and mwimonitornotify. These options allow
3523 you to enable MWI monitoring on FXO lines. When the MWI state changes,
3524 the script specified in the mwimonitornotify option is executed. An internal
3525 event indicating the new state of the mailbox is also generated, so that
3526 the normal MWI facilities in Asterisk work as usual.
3527 * Added signalling type 'auto', which attempts to use the same signalling type
3528 for a channel as configured in DAHDI. This is primarily designed for analog
3529 ports, but will also work for digital ports that are configured for FXS or FXO
3530 signalling types. This mode is also the default now, so if your chan_dahdi.conf
3531 does not specify signalling for a channel (which is unlikely as the sample
3532 configuration file has always recommended specifying it for every channel) then
3533 the 'auto' mode will be used for that channel if possible.
3534 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
3535 state for a channel; also ensured that the DNDState Manager event is
3536 emitted no matter how the DND state is set or cleared.
3540 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
3541 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
3542 for details. This new channel driver allows you to use Nortel i2002,
3543 i2004, and i2050 phones with Asterisk.
3544 * Added a new channel driver, chan_console, which uses portaudio as a cross
3545 platform audio interface. It was written as a channel driver that would
3546 work with Mac CoreAudio, but portaudio supports a number of other audio
3547 interfaces, as well. Note that this channel driver requires v19 or higher
3548 of portaudio; older versions have a different API.
3552 * Added the ability to specify arguments to the Dial application when using
3553 the DUNDi switch in the dialplan.
3554 * Added the ability to set weights for responses dynamically. This can be
3555 done using a global variable or a dialplan function. Using the SHELL()
3556 function would allow you to have an external script set the weight for
3558 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
3559 functions will allow you to initiate a DUNDi query from the dialplan,
3560 find out how many results there are, and access each one.
3561 * Added the ability to specifiy a port for a dundi peer.
3565 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
3566 functions will allow you to initiate an ENUM lookup from the dialplan,
3567 and Asterisk will cache the results. ENUMRESULT can be used to access
3568 the results without doing multiple DNS queries.
3572 * Added the ability to customize which sound files are used for some of the
3573 prompts within the Voicemail application by changing them in voicemail.conf
3574 * Added the ability for the "voicemail show users" CLI command to show users
3575 configured by the dynamic realtime configuration method.
3576 * MWI (Message Waiting Indication) handling has been significantly
3577 restructured internally to Asterisk. It is now totally event based
3578 instead of polling based. The voicemail application will notify other
3579 modules that have subscribed to MWI events when something in the mailbox
3581 This also means that if any other entity outside of Asterisk is changing
3582 the contents of mailboxes, then the voicemail application still needs to
3583 poll for changes. Examples of situations that would require this option
3584 are web interfaces to voicemail or an email client in the case of using
3585 IMAP storage. So, two new options have been added to voicemail.conf
3586 to account for this: "pollmailboxes" and "pollfreq". See the sample
3587 configuration file for details.
3588 * Added "tw" language support
3589 * Added support for storage of greetings using an IMAP server
3590 * Added ability to customize forward, reverse, stop, and pause keys for message playback
3591 * SMDI is now enabled in voicemail using the smdienable option.
3592 * A "lockmode" option has been added to asterisk.conf to configure the file
3593 locking method used for voicemail, and potentially other things in the
3594 future. The default is the old behavior, lockfile. However, there is a
3595 new method, "flock", that uses a different method for situations where the
3596 lockfile will not work, such as on SMB/CIFS mounts.
3597 * Added the ability to backup deleted messages, to ease recovery in the case
3598 that a user accidentally deletes a message, and discovers that they need it.
3599 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
3600 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
3601 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
3602 voicemail boxes. The SMDI interface can also poll for MWI changes when some
3603 outside entity is modifying the state of the mailbox (such as IMAP storage or
3604 a web interface of some kind).
3605 * Added the support for marking messages as "urgent." There are two methods to accomplish
3606 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
3607 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
3608 the message as urgent after he has recorded a voicemail by following the voice instructions.
3609 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
3614 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
3615 used across multiple queues.
3616 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
3617 setqueueentryvar options for each queue, see queues.conf.sample for details.
3618 * Added keepstats option to queues.conf which will keep queue
3619 statistics during a reload.
3620 * setinterfacevar option in queues.conf also now sets a variable
3621 called MEMBERNAME which contains the member's name.
3622 * Added 'Strategy' field to manager event QueueParams which represents
3623 the queue strategy in use.
3624 * Added option to run macro when a queue member is connected to a caller,
3625 see queues.conf.sample for details.
3626 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
3627 does not count paused queue members as unavailable.
3628 * Added min-announce-frequency option to queues.conf which allows you to control the
3629 minimum amount of time between queue announcements for use when the caller's queue
3630 position changes frequently.
3631 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
3633 * Added ability for non-realtime queues to have realtime members
3634 * Added the "linear" strategy to queues.
3635 * Added the "wrandom" strategy to queues.
3636 * Added new channel variable QUEUE_MIN_PENALTY
3637 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
3638 rules in queuerules.conf. See configs/queuerules.conf.sample for details
3639 * Added a new parameter for member definition, called state_interface. This may be
3640 used so that a member may be called via one interface but have a different interface's
3641 device state reported.
3642 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
3643 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
3644 "manager show command QueueReset."
3645 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
3646 specified by the periodic-announce option, then one will be chosen randomly when it is time
3647 to play a periodic announcment
3648 * New configuration options: announce-position now takes two more values in addition to "yes" and
3649 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
3650 announce-position-limit. By setting announce-position to "limit" callers will only have their
3651 position announced if their position is less than what is specified by announce-position-limit.
3652 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
3653 will be told that their are more than announce-position-limit callers waiting.
3654 * Two new queue log events have been added. An ADDMEMBER event will be logged
3655 when a realtime queue member is added and a REMOVEMEMBER event will be logged
3656 when a realtime queue member is removed. Since there is no calling channel associated
3657 with these events, the string "REALTIME" is placed where the channel's unique id
3658 is typically placed.
3659 * The configuration method for the "joinempty" and "leavewhenempty" options has
3660 changed to a comma-separated list of methods of determining member availability
3661 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
3662 values are still accepted for backwards-compatibility, though.
3663 * The average talktime is now calculated on queues. This information is reported via the
3664 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
3665 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
3670 * The 'o' option to provide an optimization has been removed and its functionality
3671 has been enabled by default.
3672 * When a conference is created, the UNIQUEID of the channel that caused it to be
3673 created is stored. Then, every channel that joins the conference will have the
3674 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
3675 callers that come and go from long standing conferences.
3676 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
3677 except it does operations on a channel by name, instead of number in a conference.
3678 This is a very useful feature in combination with the 'X' option to ChanSpy.
3679 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
3681 * Added new RealTime functionality to provide support for scheduled conferencing.
3682 This includes optional messages to the caller if they attempt to join before
3683 the schedule start time, or to allow the caller to join the conference early.
3684 Also included is optional support for limiting the number of callers per
3685 RealTime conference.
3686 * Added the S() and L() options to the MeetMe application. These are pretty
3687 much identical to the S() and L() options to Dial(). They let you set
3688 timeouts for the conference, as well as have warning sounds played to
3689 let the caller know how much time is left, and when it is running out.
3690 * Added the ability to do "meetme concise" with the "meetme" CLI command.
3691 This extends the concise capabilities of this CLI command to include
3692 listing all conferences, instead of an addition to the other sub commands
3693 for the "meetme" command.
3694 * Added the ability to specify the music on hold class used to play into the
3695 conference when there is only one member and the M option is used.
3696 * Added MEETME_INFO dialplan function which provides a way to query
3697 various properties of a Meetme conference.
3698 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
3699 and *84: record in-conf
3701 Other Dialplan Application Changes
3702 ----------------------------------
3703 * Argument support for Gosub application
3704 * From the to-do lists: straighten out the app timeout args:
3705 Wait() app now really does 0.3 seconds- was truncating arg to an int.
3706 WaitExten() same as Wait().
3707 Congestion() - Now takes floating pt. argument.
3708 Busy() - now takes floating pt. argument.
3709 Read() - timeout now can be floating pt.
3710 WaitForRing() now takes floating pt timeout arg.
3711 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
3712 * Added 's' option to Page application.
3713 * Added an optional timeout argument to the Page application.
3714 * Added 'E', 'V', and 'P' commands to ExternalIVR.
3715 * Added 'o' and 'X' options to Chanspy.
3716 * Added a new dialplan application, Bridge, which allows you to bridge the
3717 calling channel to any other active channel on the system.
3718 * Added the ability to specify a music on hold class to play instead of ringing
3719 for the SLATrunk application.
3720 * The Read application no longer exits the dialplan on error. Instead, it sets
3721 READSTATUS to ERROR, which you can catch and handle separately.
3722 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
3723 of asking for verification of each name, one at a time.
3724 * Privacy() no longer uses privacy.conf, as all options are specifyable as
3725 direct options to the app.
3726 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
3728 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
3729 * The ChannelRedirect application no longer exits the dialplan if the given channel
3730 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
3731 or NOCHANNEL if the given channel was not found.
3732 * The silencethreshold setting that was previously configurable in multiple
3733 applications is now settable globally via dsp.conf.
3735 Music On Hold Changes
3736 ---------------------
3737 * A new option, "digit", has been added for music on hold classes in
3738 musiconhold.conf. If this is set for a music on hold class, a caller
3739 listening to music on hold can press this digit to switch to listening
3740 to this music on hold class.
3741 * Support for realtime music on hold has been added.
3742 * In conjunction with the realtime music on hold, a general section has
3743 been added to musiconhold.conf, its sole variable is cachertclasses. If this
3744 is set, then music on hold classes found in realtime will be cached in memory.
3748 * AEL upgraded to use the Gosub with Arguments instead
3749 of Macro application, to hopefully reduce the problems
3750 seen with the artificially low stack ceiling that
3751 Macro bumps into. Macros can only call other Macros
3752 to a depth of 7. Tests run using gosub, show depths
3753 limited only by virtual memory. A small test demonstrated
3754 recursive call depths of 100,000 without problems.
3755 -- in addition to this, all apps that allowed a macro
3756 to be called, as in Dial, queues, etc, are now allowing
3757 a gosub call in similar fashion.
3758 * AEL now generates LOCAL(argname) declarations when it
3759 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
3760 etc. That makes the arguments local in scope. The user
3761 can define their own local variables in macros, now,
3762 by saying "local myvar=someval;" or using Set() in this
3763 fashion: Set(LOCAL(myvar)=someval); ("local" is now
3765 * utils/conf2ael introduced. Will convert an extensions.conf
3766 file into extensions.ael. Very crude and unfinished, but
3767 will be improved as time goes by. Should be useful for a
3768 first pass at conversion.
3769 * aelparse will now read extensions.conf to see if a referenced
3770 macro or context is there before issueing a warning.
3771 * AEL parser sets a local channel variable ~~EXTEN~~, to
3772 preserve the value of ${EXTEN} thru switch statements.
3773 * New operator in $[...] expressions: the ~~ operator serves
3774 as a concatenation operator. AT THE MOMENT, it is really only
3775 necessary and useful in AEL, especially in if() expressions.
3776 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
3777 any enclosing double-quotes, and evaluate to the value of a
3778 concatenated with the value of b. For example if a is set to
3779 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
3780 evaluate to xyzabc .
3783 Call Features (res_features) Changes
3784 ------------------------------------
3785 * Added the parkedcalltransfers option to features.conf
3786 * Added parkedcallparking option to control one touch parking w/ parking
3788 * Added parkedcallhangup option to control disconnect feature w/ parking
3790 * Added parkedcallrecording option to control one-touch record w/ parking
3792 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
3793 parkedcalltransfers option support for multiple parking lots.
3794 * Added BRIDGE_FEATURES variable to set available features for a channel
3795 * The built-in method for doing attended transfers has been updated to
3796 include some new options that allow you to have the transferee sent
3797 back to the person that did the transfer if the transfer is not successful.
3798 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
3799 in features.conf.sample.
3800 * Added support for configuring named groups of custom call features in
3801 features.conf. This means that features can be written a single time, and
3802 then mapped into groups of features for different key mappings or easier
3804 * Updated the ParkedCall application to allow you to not specify a parking
3805 extension. If you don't specify a parking space to pick up, it will grab
3806 the first one available.
3807 * Added cli command 'features reload' to reload call features from features.conf
3808 * Moved into core asterisk binary.
3809 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
3810 * Added the ability for custom parking lots to be configured with their own
3811 parking extension with the parkext option.
3813 Language Support Changes
3814 ------------------------
3815 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
3816 * Added support for the Hungarian language for saying numbers, dates, and times.
3820 * Added SPEECH commands for speech recognition. A complete listing can be found
3822 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
3823 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
3824 does not behave as expected; the native command needs to be used, instead.
3825 * Added the ability to perform SRV lookups on fast AGI calls. To use this
3826 feature, simply use hagi: instead of agi: as the protocol portion
3827 of the URI parameter to the AGI function call in your dial plan. Also note
3828 that specifying a port number in the AGI URI will disable SRV lookups,
3829 even if you use the hagi: protocol.
3830 * No longer support MSG_OOB flag on HANGUP.
3834 * Added rotatestrategy option to logger.conf, along with two new options:
3835 "timestamp" which will use the time to name the logger files instead of
3836 sequence number; and "rotate", which rotates the names of the log files,
3837 similar to the way syslog rotates files.
3838 * Added exec_after_rotate option to logger.conf, which allows a system
3839 command to be run after rotation. This is primarily useful with
3840 rotatestrategy=rotate, to allow a limit on the number of log files kept
3841 and to ensure that the oldest log file gets deleted.
3842 * Added realtime support for the queue log
3846 * The cdr_manager module has a [mappings] feature, like cdr_custom,
3847 to add fields to the manager event from the CDR variables.
3848 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
3849 backend database CDR table. Specifically, additional, non-standard
3850 columns are supported, merely by setting the corresponding CDR variable in
3851 your dialplan. In addition, you may alias any column to another name (for
3852 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
3853 simply "alias src => ANI" in the configuration file). Records may be
3854 posted to more than one backend, simply by specifying multiple categories
3855 in the configuration file. And finally, you may filter which CDRs get
3856 posted to each backend, by specifying a filter (which the record must
3857 match) for the particular category. Filters are additive (meaning all
3858 rules must match to post that CDR).
3859 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
3860 module. Specifically, you may add additional columns into the table and
3861 they will be set, if you set the corresponding CDR variable name. Also,
3862 if you omit columns in your database table, they will be silently skipped
3863 (but a record will still be inserted, based on what columns remain). Note
3864 that the other two features from cdr_adaptive_odbc (alias and filter) are
3865 not currently supported.
3866 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
3867 has been disabled using the NoCDR application.
3869 Miscellaneous New Modules
3870 -------------------------
3871 * Added a new CDR module, cdr_sqlite3_custom.
3872 * Added a new realtime configuration module, res_config_sqlite
3873 * Added a new codec translation module, codec_resample, which re-samples
3874 signed linear audio between 8 kHz and 16 kHz to help support wideband
3876 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
3877 based on configuration templates that use Asterisk dialplan function and
3878 variable substitution. It should be possible to create phone profiles and
3879 templates that work for the majority of phones provisioned over http. It
3880 is currently only intended to provision a single user account per phone.
3881 An example profile and set of templates for Polycom phones is provided.
3882 NOTE: Polycom firmware is not included, but should be placed in
3883 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
3884 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
3885 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
3886 provided; there is a JACK() application, and a JACK_HOOK() function. Both
3887 interfaces create an input and output JACK port. The application makes
3888 these ports the endpoint of the call. The audio coming from the channel
3889 goes out the output port and whatever comes back in on the input port is
3890 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
3891 audiohook on the channel. This lets you run the audio coming from a
3892 channel through JACK, and whatever comes back in is what gets forwarded
3893 on as the channel's audio. This is very useful for building custom
3894 vocoders or doing recording or analysis of the channel's audio in another
3896 * Added a new module, res_config_curl, which permits using a HTTP POST url
3897 to retrieve, create, update, and delete realtime information from a remote
3898 web server. Note that this module requires func_curl.so to be loaded for
3899 backend functionality.
3900 * Added a new module, res_config_ldap, which permits the use of an LDAP
3901 server for realtime data access.
3902 * Added support for writing and running your dialplan in lua using the pbx_lua
3903 module. See configs/extensions.lua.sample for examples of how to do this.
3907 * Ability to use libcap to set high ToS bits when non-root
3908 on Linux. If configure is unable to find libcap then you
3909 can use --with-cap to specify the path.
3910 * Added maxfiles option to options section of asterisk.conf which allows you to specify
3911 what Asterisk should set as the maximum number of open files when it loads.
3912 * Added the jittertargetextra configuration option.
3913 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
3914 configuration files for the IP channel drivers. The new option is "cos".
3915 This information is also documented on the Asterisk wiki at
3916 https://wiki.asterisk.org/wiki/x/EYBG
3917 * When originating a call using AMI or pbx_spool that fails the reason for failure
3918 will now be available in the failed extension using the REASON dialplan variable.
3919 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
3920 It allows you to configure a prefix for auto-monitor recordings.
3921 * A new extension pattern matching algorithm, based on a trie, is introduced
3922 here, that could noticeably speed up mid-sized to large dialplans.
3923 It is NOT used by default, as duplicating the behaviour of the old pattern
3924 matcher is still under development. A config file option, in extensions.conf,
3925 in the [general] section, called "extenpatternmatchingnew", is by default
3926 set to false; setting that to true will force the use of the new algorithm.
3927 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
3928 be used to switch the algorithms at run time.
3929 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
3930 specifying which socket to use to connect to the running Asterisk daemon
3932 * Performance enhancements to the sched facility, which is used in
3933 the channel drivers, etc. Added hashtabs and doubly-linked lists
3934 to speed up deletion; start at the beginning or end of list to
3936 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
3937 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
3938 Added regression tests to the tests/ dir, also.
3939 * Added a refcount trace feature to astobj2 for those trying to balance
3940 object creation, deletion; work, play; space and time. See the
3941 notes in astobj2.h. Also, see utils/refcounter as well, as a
3942 quick way to find unbalanced refcounts in what could be a sea
3943 of objects that were balanced.
3944 * Added logging to 'make update' command. See update.log
3945 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
3946 do not come from the remote party.
3947 * Added the 'n' option to the SpeechBackground application to tell it to not
3948 answer the channel if it has not already been answered.
3949 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
3950 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
3952 * iLBC source code no longer included (see UPGRADE.txt for details)
3953 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
3954 deadlock is detected, a backtrace of the stack which led to the lock calls
3955 will be output to the CLI.
3956 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
3957 the "core show locks" CLI command will give lock information output as well
3958 as a backtrace of the stack which led to the lock calls.
3959 * users.conf now sports an optional alternateexts property, which permits
3960 allocation of additional extensions which will reach the specified user.
3961 * A new option for the configure script, --enable-internal-poll, has been added
3962 for use with systems which may have a buggy implementation of the poll system
3963 call. If you notice odd behavior such as the CLI being unresponsive on remote
3964 consoles, you may want to try using this option. This option is enabled by default
3965 on Darwin systems since it is known that the Darwin poll() implementation has
3969 --------------------
3970 * In addition to timing from DAHDI, there is a new timing module called
3971 res_timing_timerfd. In order to use this, you must be running Linux with
3972 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
3973 script will be able to tell if you have the requirements. From menuselect, select
3974 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob